摘要:
An apparatus and method that determines the end-to-end transit delay at each node of a path, in accordance with the selected probability value indicative of the probability to experience a delay at each node that is smaller than the computed transit delay. Then the computed transit delays per nodes are combined to obtain the end-to-end delay of the path, the combination being either an arithmetic operation or a convolution operation. A method to approximate the convolution operation is also disclosed.
摘要:
This method enables optimizing the time required for reestablishing connections between end users attached to a data communication network, which connections were disrupted due to a network failure. The network includes access nodes and transit nodes interconnected with network links/trunks (with no specific distinction being herein required between both designations of a communication line). The end users are attached to the network through access nodes and each said access node permanently stores an image of the current network trunk including the number N.sub.i of connections currently supported by said network trunk. Upon detection of a trunk failure, each access node supporting connections affected by said failure, is made aware of the total number (N.sub.i) of connections in each priority group affected by the failure, together with a network dependent parameter (TP) representing the elementary processing time required to reroute a single network connection. Then each access node may independently, start a first reconnection set-up procedure at a time Ri randomly selected between zero and (N.sub.i -n.sub.i) *TP, where n.sub.i is the number of connections supported by the access node in each priority group, and then space the required subsequent reconnections set-ups by a time equal to (T.sub.i -R.sub.i)/n.sub.i, with T.sub.i =N.sub.i *TP.
摘要:
A method based on predefined connection priorities for assigning link bandwidth to a requesting user in a high speed digital network interconnecting network users through a path including network nodes connected through high speed links. According to this method, a predefined reservable link bandwidth is split into so-called nominal bandwidth portions and common bandwidth portions, both assignable to the same connections on a priority basis. Each of the common bandwidth priorities is individually related to a nominal bandwidth priority through a predefined relationship, making the common bandwidth priorities always lower than any nominal priority. In this way the requested link connection bandwidth, whatever be its nominal priority, is made preemptable primarily on all common bandwidth, thus avoiding the disruption of any network connection which is already established.
摘要:
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.
摘要:
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.
摘要:
A process for multirate subband encoding voice signals.At least a portion of the original signal bandwidth is split into p sub-bands the contents of which are to be coded using dynamic allocation of quantizing levels throughout the sub-bands.Let's assume the signal is to be coded for a set of coding rates R(1)
摘要:
A method of reconstructing, at the receiving end, digital data defining an encoded voice signal segment lost in transmission between the transmitter and the receiver in a transmission system wherein a low-frequency signal such as a residual baseband signal is derived at the transmitting end from the signal to be encoded and is then distributed among several sub-bands whose sampled contents are quantized separately. The reconstruction method includes a step of analyzing the received signal to detect any missing segment thereof and, as the case may be, to initiate an analysis, within each sub-band, of the segment(s) adjacent to the lost segment, so as to generate a term relating to the period T.sub.(k) of the signal so analyzed and to reconstruct a segment of signal of period T.sub.(k) intended to be substituted for the lost segment. Cyclic redundancy is used to detect errors, and modulo sequence numbering identification is used to detect loss of packet.
摘要:
This coder dispatches the bits resulting from the coding operation of a speech signal into a format for transmission at a rate chosen from a plurality of predetermined transmission rates. More specifically, the contents of at least part of the speech signal frequency bandwidth is split into several sub-bands. Said sub-bands are regrouped into sub-groups, each of said sub-groups corresponding to at least one of said possible transmission rates, i.e. to at least one sub-group coding bit rate. The signal samples belonging to each sub-group are recoded through a dynamical allocation of requantizing steps between the sub-bands. The obtained bits resulting from the recoding are dispatched into a multirate frame according to the sub-group which they belong to.
摘要:
This improved speech signal Block Coded PCM (BCPCM) system reduces the number of bits allocated to transmitting the scale factor, thereby releasing bits for allocation to coding samples in the associated block of samples. The scale factor (c) is calculated for every 16 millisecond block of samples. However, the scale factor will be transmitted only once per 32 millisecond block if there is no significant difference between the two sequential values. The original speech signal is split into 16 frequency subbands, each subband initials sampled and 12-bit coded, then requantized in BCPCM at dynamically variable bit rates depending on the scale factor transmission rate.
摘要:
A coding process, given a total number of available bits per unit time, allocates to each of a plurality of signal channels a number of bits proportional to information in the channel. The process is useful for transcoding BCPCM (Block Coded PCM) voice signals for lower bit rate and conserving transmission channel bandwidth.