http://rdf.ncbi.nlm.nih.gov/pubchem/patent/CA-1273697-A

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filingDate 1986-08-22-04:00^^<http://www.w3.org/2001/XMLSchema#date>
grantDate 1990-09-04-04:00^^<http://www.w3.org/2001/XMLSchema#date>
inventor http://rdf.ncbi.nlm.nih.gov/pubchem/patentinventor/MD5_cf74245a14fd09c11c360ca33faaef51
publicationDate 1990-09-04-04:00^^<http://www.w3.org/2001/XMLSchema#date>
publicationNumber CA-1273697-A
titleOfInvention Multiplexed digital packet telephone system
abstract MULTIPLEXED DIGITAL PACKET TELEPHONE SYSTEM ABSTRACT An efficient system for simultaneously conveying a large number of telephone conversations over a much smal-ler number of relatively low-bandwidth digital communication channels is disclosed. Each incoming telephone speech signal is filtered, periodically sampled, and digitized. An efficient computational speech compression algorithm is applied to transform a sequence of digitized speech samples into a much shorter sequence of compression vari-ables. The compression variable sequence is further pro-cessed to construct a minimum-length bit string, and an identifying header is appended to form a packet. Only a few packets containing information on representative background noise are generated during pauses in speech, thereby conserving digital bandwidth. The packets are queued and transmitted asynchronously over the first avail-able serial digital communication channel. Numerical feed-back to the compression algorithm is employed which results in the packet size being reduced during periods of high digital channel usage. Packet header information is uti-lized to establish a "virtual circuit" between sender and receiver. At the ultimate receiving terminal, the packet header is employed to route the packet to the circuitry servicing the appropriate telephone channel. The packeting procedure is thereupon reversed. The header is stripped from the packet and the minimum length bit string is expand-ed to reproduce the original sequence of compression vari-ables. By means of an appropriate computational inverse speech compression algorithm, approximations to the original digital samples are synthesized. An intentional delay is thereupon introduced to build a backlog pool of samples to buffer against gaps in speech due to statistical fluctua-tions in packet arrival times. Digital samples from this pool are periodically outputted to a D/A converter and its analog output is appropriately filtered to approximately reproduce the original frequency-limited telephone signal. Representative background noise is synthesized from informa-tion contained in previously received background packets during pauses in speech.
priorityDate 1985-08-23-04:00^^<http://www.w3.org/2001/XMLSchema#date>
type http://data.epo.org/linked-data/def/patent/Publication

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