Abstract: A method and an apparatus implementing a shaping function in a fixed length cell switching network node adapter supporting output lines in a telecommunication network. The adapter includes a Time Division Multiplexing (TDM) table where each entry is a pointer to a first cell of a queue of cells stored in the adapter for a given traffic. When a new traffic is established, the TDM table is updated. A placement device performs the pre-computation of the best placement of the entries reserved for one traffic in the TDM table. The placement device minimizes the Cell Delay Variation as computed by a Generic Cell Rate Algorithm (GCRA--ITU standard organization) policer. The Cell Delay Variation (CDV) is defined as the deviation of the placement of the cells from their ideal position in an output stream, the ideal placement being when the cells are spaced with an interval corresponding to the period of traffic negotiated at traffic establishment time.

Abstract: Voice circuit emulation system in a packet switching network includes a plurality of switching nodes (SW-1, SW-2, . . . ) interconnected by connection lines (TL-1, TL-2, . . . ), enabling voice signals to be transmitted in circuit emulation packets from any source exchange telephone device (PBX-A) to any destination exchange telephone device (PBX-B). Each switching node (SW-1 or SW-2) includes a circuit emulation server (CES-1 or CES-2) including a plurality of connection tables corresponding each to an incoming connection line, each table containing for each incoming connection line, the identification of each outgoing connection line associated to each slot of the circuit emulation packet received from the incoming connection line and the identification of each slot of the packet to be transmitted on the outgoing connection line.

Abstract: A multipurpose data communication network node for interconnecting both ATM and Variable Length (VL) input/output trunks with all combinations of node input versus output trunk connections. The network node includes ATM/VL Receive Adapters and ATM/VL Transmit Adapters, interconnected via a Switching device (44) operating on ATM like packets (i.e. ATM cells) only. The receive adapter includes means (41, 43) for deriving switchable cells from VL traffic possibly including ATM packets and provided over an input VL trunk and means (45, 46) for deriving switchable cells from ATM packets provided on input ATM trunk. The transmit adapter, includes means (47, 48) for reconstructing VL traffic to be fed onto an output VL trunk, and means (49, 50) for reconstructing ATM traffic to be fed onto an output ATM trunk; both means (47, 48) and (49, 50) being fed with switchable cells irrespective of the traffic origin, being it from VL or ATM trunks.

Abstract: An apparatus and method for multicasting messages stored in data buffers of a data storage. Each message is composed of data stored in a plurality of the data buffers. Each data buffer is controlled and mapped to a unique direct control block (DCB) which stores information characterizing the data buffer. By chaining the DCBs variable length, messages can be generated. Indirect control blocks (ICB) stores information characterizing the data or messages duplicated and points to a DCB. A field in the DCB carries a count representing the number of times the message is to be duplicated.

Abstract: In a packet switched communications system an incoming real-time packet is imbedded after the next block of data of the non-real-time packet being transmitted. This object is accomplished by transmitting each packet along with at least a 1-byte trailer which is used to indicate the packet type, whether the current block of non real time data is preempted or whether the current block of non real time data is resumed.

Abstract: A high performance data packet buffering method and a programmable data communication adapter for high speed packet transmission networks are disclosed. The line adapter includes programmable processing means, for receiving and transmitting data packets of fixed or variable length. This system is characterized in that it comprisesmeans for buffering (132) said data packets,means for identifying said buffering means and said data packets in said buffering means,means for queueing (FIG. 15) in storing means (131) said identifying means in a single instruction,means for dequeueing (FIG. 16) from said storing (131) means said identifying means in another single instruction,means for releasing said buffering means,Each instruction comprises up to three operations executed in parallel by said processing means:an arithmetical and logical (ALU) operation on said identifying means,memory operation on said storing means, anda sequence operation.

Abstract: Currently, routing algorithms compute all the available paths in the network, from a source node to a destination node before selecting the optimal route. The route computation is often time and resource consuming. Some paths are not acceptable due to the particular geographical configuration of the network. In the real world, large transport networks are not fully meshed. The present invention is based on the observation that networks are usually built around a hierarchical structure. A set of nodes, interconnected by high throughput lines, are used to build a `Backbone` (401) with a high degree of meshing to allow the redundancy and reliability required by the user. The other nodes or `local` nodes (404) are attached to one or several backbone nodes. It is the network designer responsibility, at the configuration time to define for each node what is its attribution: backbone (402) or local node (404).

Abstract: A packet switching communication system is improved by using a packet header structure which does not require a fixed format. The packet header comprises a chain of 2 byte command/data segments. Each command/data segment contains generic bits and a routing field. One of the generic bits (bit 1) allows the header to be extended with another command/data segment.

Abstract: Access control for a packet communications network includes a dynamic bandwidth updating mechanism which continuously monitors the mean bit rate of the signal source and the loss probability of the connection. These values are filtered to remove noise and then used to test whether the values fall within a pre-defined acceptable adaptation region in the mean bit rate, loss probability plane. Values falling outside of this region trigger bandwidth updating procedures which, in turn, result in acquiring a new connection bandwidth, and determining new filter parameters and new parameters for a leaky bucket access mechanism.

Abstract: A DTMF tone is detected through the tracking of two tunable filters HAVING coefficients within so-called tunnels each limited by thresholds derived from the tone frequency components to be detected. In and out-band energies E1(n) and E2(n) are used to further validate tone detection.

Abstract: In a voice coding system, the baseband or residual signal is encoded at a lower rate by finding a best estimate at a lower rate. The voice terminal signal x(n) is split into a low-pass filtered band signal y1(n) and a high-pass filtered band signal y2(n). Both y1(n) and y2(n) signals are coded into lower-rate sub-sequences of samples x1(n), x2(n) and x3(n), x4(n) respectively. The sequence of samples to be representative of x(n) is selected among x1(n), x2(n), x3(n) and x4(n) for being the closest to x(n).

Abstract: An improved digital signal processor (DSP) architecture includes several multiply/accumulate devices (M-Unit 0-K) connected to the DSP bus through a delay line and elements for simultaneously operating the multiply/accumulate devices and a device for selectively storing accumulated values into a pre-assigned Dual-Port Randomly Accessible Memory area.

Abstract: A vector quantizing speech coder includes a Short-Term-Predictive filter which receives originally sampled speech signal s(n) and decorrelates it into a residual signal r(n). A device is provided to quantize the residual signal r(n) at a low bit rate. The device also generates a reconstructed signal r(n) from which coefficients for adjusting the Short-Term-Predictive filter are dynamically derived.

Abstract: A digital filter providing output samples at a rate B while input samples are being provided at a rate A. The filter impulse response h(t) is sampled at a rate P=a.A=b.B, with P being the lowest common multiple to A and B, to derive Np filter coefficients therefrom. The coefficients are arranged into "a" sets of Np/a coefficients and stored. Output samples are then computed by feeding input samples Sn at incoming rate while permutating the sets of coefficients and generating output samples at the output rate.

Abstract: A process for deriving voice pitch related delay values M to tune a Long-Term Prediction (LTP) filter to be used in an LTP based speech coder converting a speech derived digital signal r(n) into a lower bit rate signal, said filter being provided with a variable length delay line y(n) fed with a reconstructed signal r'(n). The process includes splitting r(n) into segments and each segment into sub-segments; then cross-correlating the first current r(n) sub-segment with a previously reconstructed segment and sorting the cross-correlation values for peak location, whereby a first delay value M1 is derived and used to tune the filter. Then, said M1 is used to compute sample indexes n for a predefined number of samples located about M1/p, . . . , M1, 2M1, . . . , pM1 and repeating cross-correlation and sorting operations to derive M2 and so on up to a full segment length (e.g. 160 samples). Then the process is started all over again.

Abstract: The process for varying the speed of a speech signal that involves splitting at least a portion of the speech frequency bandwidth into N narrow sub-bands, processing each sub-hand signal contents to derive therefrom magnitude data M(i, n) and phase data P(i, n), i=1, . . . , N being the subband index and n the time index. The M (i, n) sequence is converted into a sequence M'(n) by either duplicating one sample every K samples (K being an integer value derived from the desired slowing-down/speeding up ratio). The phase sequence P (i, n) is processed to derive therefrom an increment sequence D(i, n)=P(i, n)-P(i, n-1), which increment sequence is first converted into a D'(i, n) sequence by either dropping or duplicating one sample every K, samples, before being converted into P'(i, n)=P'(i, n)+D'(i, n). The P'(i, n), D'(i, n) sequences are converted back into sub-band signals contents, then combined together into the slowed-down/speeded-up speech signal.

Abstract: Some signal coding techniques involve quantizing the signal samples using a signal derived parameter to set the quantizer step. This parameter is then used in the decoding process to synthesize the original signal back to its analog form. Any error affecting the integrity of said parameter, between coding and decoding, would affect the synthesized signal. The method proposed here for protecting data integrity involves, at the encoding site, forcing the parameter to a fixed parity. Then, before synthesizing operations are performed, any parameter failing a parity control test is discarded and replaced with a parameter derived from available valid parameter(s).

Abstract: A method for dividing a message into a plurality of message blocks and storing each of the plurality of message blocks on separate units within a distributed data storage system. Facilities for retrieving the information from the distributed data storage system and recreating the original message for replay are also provided. If a unit within the distributed data storage system fails, then a degredated version of the original message is recreated from the remaining units and their stored message blocks.

Abstract: A signal vector quantizing coder (CELP) is provided with an adaptive codebook originally loaded with preselected codewords. The codebook is split into a fixed contents portion and a fixed length adaptive contents portion. During coding operations, the codewords dynamically selected for coding the coder input signal are shifted into the fixed length adaptive codebook section for codebook contents updating purposes.

Abstract: The voice signal is analyzed to derive therefrom a low frequency base band signal, linear prediction coefficients and high frequency (HF) descriptors. Said HF descriptors include HF energy indications as well as indications relative to the phase shift between the low frequency and the high frequency band. Said HF descriptors are used during the voice synthesis operation to provide an inphase HF bandwidth component to be added to the base band prior to be used for driving a linear prediction synthesis filter tuned using said linear prediction parameters.