Abstract: An extended impulse response characteristic is obtained by connecting a plurality of adaptive echo cancelers in a tandem connection (FIG. 1 or FIG. 2). Delay units (102) are associated with receive inputs (X) of the adaptive echo cancelers (101, FIG. 1, or 201 FIG. 2) so that each canceler models a predetermined fixed portion of the extended impulse response characteristic. The relative positions of the cancelers in the tandem connection are controllably switched (via 103-0 through 103-(N-1) and (104) at a predetermined rate (via 105) to insure so-called "good" adaptation of the individual adaptive echo cancelers. In one embodiment, the adaptive echo canceler in a preferred position in the tandem connection is allowed to adapt at its normal rate while the other cancelers are inhibited (via INH) from adapting.

Abstract: In a multilocation conference terminal for transmitting audio and video information to a plurality of remote locations, rapid video and non-disrupted audio switching are obtained by employing a plurality of single composite digital signals each including all the video information from an associated location and all of the audio mixes for all of the locations in a conference. Then, the composite signals are switched prior to conversion for transmission to one or more of the locations. Since each composite signal includes all the audio mixes for all of the locations, when video from anyone of the locations is selected for transmission to any of the locations, the audio mixes for those locations are also switched.

Abstract: An estimate of delay (.DELTA.) in a communication circuit is obtained by subsampling (via 110-114) signals incoming (X(K)) to and outgoing (Y(K)) from the communication circuit, obtaining a cross correlation (via 120, 121) of the subsampled signals and processing (via 122) the cross correlation in accordance with a prescribed algorithm.In one application (FIG. 1) the delay estimate (.DELTA.) is used in conjunction with an adjustable delay (101) connected in series with a "short" echo canceler (102) across the communication circuit for cancelling echoes.

Abstract: Audio mixes for a plurality of conference locations are controllably obtained by using a matrix multiplication arrangement. The audio mixes for the individual conference locations are obtained by multiplying a dynamically determined gain matrix by an audio input matrix. Coefficient values in the gain matrix are dependent on whether incoming and outgoing transmission paths are assigned to a location and on whether an audio sample is being received or not.

Abstract: An extended impulse response characteristic is obtained by switchably connecting (via 13, 14, 15) at least first (10) and second (12) adaptive echo cancelers in a tandem connection (FIG. 1). A delay unit (11) is associated with a receive input (X) of the second adaptive echo canceler (12) so that each filter models a different portion of the extended impulse response characteristic. The relative positions of the adaptive echo cancelers (10, 12) in the tandem connection are controllably switched (via 13, 14, 15 and 16) to insure so-called "good" adaptation of the individual adaptive echo cancelers (10, 12) to minimize misalignment noise. Both adaptive echo cancelers (10, 12) are allowed to adapt initially but upon switching only the adaptive echo canceler (12) in a so-called preferred position in the tandem connection is allowed to adapt while the other adaptive echo canceler (10) is inhibited from further adapting.

Abstract: An input signal sample is converted, i.e., quantized, from a first digital code to a second digital code by combining an array of stored differential quantizer decision levels with the input signal sample to obtain an array of partial sum signal values. The signs of the partial sum signal values are algebraically summed and the result is used to obtain the quantizer interval that the input signal sample falls in.

Abstract: Improved efficiency in allocating bits to individual ones of a plurality of sub-band signals (or amplitude coefficients) is realized by employing a so-called template matching technique. A plurality of the most likely to occur bit allocation patterns and a corresponding plurality of templates including representations of a prescribed characteristic of the sub-band signals that would generate a corresponding one of the bit allocation patterns are stored for later use. Selection of a bit allocation pattern for use in encoding and/or decoding the sub-band signals is effected by matching in accordance with a given one of the stored templates to representations of the prescribed characteristic of the sub-band signals presently being encoded. In one example, the prescribed characteristic is a power estimate and the matching criterion is a minimum rate distortion criterion.

Abstract: In a multilocation video conference system, contention for the video to be transmitted to the locations is resolved by employing so-called talker and graphics contention resolution processes. Both the talker and graphics contention processes may be overridden by manual selection of the video to be viewed at each location. In the talker contention process, the video to be transmitted is not switched until one and only one talker location is detected during a predetermined talker timing interval. In the graphics contention process, all requests for graphics transmission are rejected until one and only one graphics transmission request is detected during a predetermined graphics request timing interval. Once transmission of graphics has been assigned to a location, all other locations in the conference are transmitted the graphics video information until the location displaying graphics releases from the graphics transmission mode or there is a manual override.

Abstract: In a Multilocation Conference Terminal (MLCT), disruptions in the video being viewed which are caused by switching from one conference location to another are minimized by delaying switching of the video signal until a so-called get ready control signal is received from the conference location to be viewed. The conference location transmits the get ready control signal a predetermined interval prior to transmitting a so-called fast video picture update in response to a request from a Multilocation Conference Terminal (MLCT). Additional possible disruptions in the video are eliminated by minimizing the number of fast video picture updates that are used to serve a multiplicity of requests from other conference locations. When multiple MLCTs are employed in a conference, the MLCTs communicate to each other the video select modes of all the conference locations in the conference, i.e., a manual select mode or an automatic select mode.

Abstract: A modified parity calculation technique is presented which obviates the conventional requirement of deterministically inserting markers into a bit stream of data. The advantages of the present technique are fully realized in applications for in-service monitoring of extremely high speed bit streams of data. A bit pattern, or template, is compared to the content of the bit stream to detect matches between the two. The matches determine bit spans on which the parity calculation is performed. Since the parity signal is highly compressed and low speed relative to the data, it can be readily delayed to facilitate confirmation of identical content of two high speed bit streams which differ substantially in relative timing.

Abstract: In a multilocation video conference system, conference locations are dynamically added to or deleted from an ongoing conference without disrupting it. This is realized by controllably enabling or disabling communications ports assigned to the conference locations in a Multilocation Conference Terminal (MLCT). Both audio information and control channel information transmitted from a conference location assigned to a port are monitored to determine if the audio information is satisfactory and if the control channel communication is "good". If the audio information is unsatisfactory, it is muted and selection of the video from that location as a one-and-only-one talker is inhibited. If the control channel communication is not good the video signal from the conference location is excluded from the conference.

Abstract: Accumulation of distortion possible from multiple digital code conversions is eliminated in an ADPCM coder that converts PCM incoming signals, e.g., .mu.-law PCM to linear PCM and, then, to a quantized n-bit differential PCM output signal. Elimination of distortion accumulation in subsequent code conversion stages is realized in the coder when the differential PCM output is not on the most positive or most negative ADPCM quantizer steps by controllably modifying the coder differential PCM output so that a next subsequent coder would generate the same differential PCM value.Accumulation of distortion in subsequent code conversions when the differential PCM coder output is either the most positive or most negative quantizer step is eliminated by controllably modifying a decoder output so that the next subsequent code converter will choose the same ADPCM code value.

Abstract: A bit compression coding circuit incorporates signaling bit insertion. An input signal sample(s) representing, for example, PCM encoded speech or voiceband data, is delivered to a difference circuit (31) where a predicted signal (s.sub.e) is subtracted from it. The predicted signal is an estimate of the input sample derived from a predictor (32). The resultant difference signal is coupled to the input of an adaptive quantizer (34) which provides at its output a bit compressed quantized differential PCM version of the difference signal. A multiplexer (37) receives the output of the quantizer and serves to periodically preempt the least significant bit of the bit compressed PCM signal and substitute a signaling bit therefor. The output of the multiplexer is coupled to the input of an adder (38) wherein it is added to the predicted signal.

Abstract: Adaptive bit allocation to a set of spectral transform coefficients is optimized by setting a predetermined first (maximum) threshold, then allocating one bit to those coefficients above the threshold, no bits to those below. If the total number of bits has not been allocated, the threshold is lowered an "integer" step to a second threshold, and again one bit allocated to those coefficients above the second threshold. The procedure is iterated until all the total bits are allocated or, if too many bits are allocated after the last integer step, a "fractional" step threshold is used.

Abstract: Bidirectional communications between an alarm loop, comprising one or more alarm transmitters, and a remote receiver is provided through a digital transmission facility by a pair of interface units (201,301). Each interface unit, disposed at an end of the digital transmission facilities, provides encoding and decoding of signals to and from the alarm receiver. For greater reliability, each interface unit (601,701) can be adapted to select signals received from different digital transmission facilities.

Abstract: An energy discriminator is employed in conjunction with an adaptive filter to control updating of the filter transfer function characteristic. Specifically, the discriminator is employed to distinguish whether any significant received far end energy is only partial band or whole band. If the received energy is partial band the adaptive filter is inhibited from updating the transfer function characteristic during intervals that such energy is being received. On the other hand, if the received energy is determined not to be partial band and, hence, is whole band, the filter is enabled to update the transfer function characteristic during intervals that such energy is being received.In a specific example, the discriminator is employed in an adaptive echo canceler to inhibit updating an echo path estimate being generated by an adaptive transversal filter when partial band energy is being received and enabling updating of the echo path estimate when whole band energy is being received.

Abstract: Data relative to code instruction use of a central processor unit (CPU) is collected by accumulating counts of code instructions to be executed upon termination of a clock interrupt process over a predetermined interval. Information stored on occurrence of a clock interrupt pulse is evaluated to determine if the code instruction is one under evaluation. If the code instruction is one under evaluation, a count in an associated memory location is incremented. Upon termination of the predetermined interval, the data is read out in histogram form. The data in the histogram is relative to the virtual addresses of the code instructions and, therefore, directly useable by a programmer to evaluate the code use of the CPU without the need for costly and time consuming data unmapping.

Abstract: Asynchronous digital frequency shift keyed (FSK) signals are demodulated by using an autocorrelation type demodulator including digital integration. To this end, the autocorrelation function and digital integration are realized by generating integration increments, i.e., area increments, at the digital signal sampling points and accumulating them to obtain the integrated output. This is realized by turning to account a relationship between zero crossings of the received signal, zero crossings of a delayed version of the received signal and polarity of the product of the received signal and its delayed version relative to sampling points of the received digital signal. Specifically, the area increment in an n.sup.th sampling interval is determined by the time interval since the last zero crossing of the received signal before the n.sup.th sampling point, the time interval since the last zero crossing of the delayed signal before the n.sup.th sampling point and the polarity of the product signal at the n.sup.

Abstract: Accumulation of distortion possibly resulting from multiple digital code conversions is prevented by including in an ADPCM-to-PCM decoder (100, FIG. 1 or 600, FIG. 6) an arrangement (119, FIG. 1 or 601, FIG. 6) for making a trial transmission of the resulting PCM sample (.PSI.(k)) to obtain (via 209, FIG. 2) a composite noise value (.alpha.(k)) for the PCM sample (.PSI.(k)). The composite noise value includes noise components resulting from both quantizing the code word and truncating digital signals used in the decoder and subsequent coder. The composite noise value is evaluated (via 211) to determine whether adjustment of the PCM sample value (.PSI.(k)) is necessary.In one embodiment (FIG. 1, 100), a reconstructed linear version (Y.sub.r (k)) of the PCM sample is adjusted (via 216, 218, 220 or 221) when the composite noise value (.alpha.(k)) is not within the ADPCM quantizer interval for the ADPCM sample (I.sub.n (k)) being converted to PCM.

Abstract: Loop gain normalization is employed in adaptive filters to control weighting of the filter characteristic updates in order to converge properly to a desired filter characteristic. Filter stability and rapid high quality convergence is realized for a variety of received or inputted signals by employing both long term and fast attack estimates of a prescribed input signal characteristic to normalize the update gain. In one embodiment, both long term and fast attack input signal power estimates are generated and one of the two estimate values is selected to normalize the update gain. Specifically, the fast attack estimate is modified by a predetermined value and, then, the larger of the long term estimate and modified fast attack estimate is selected to normalize the update gain.