| 摘要 |
An efficient system for simultaneously conveying a large number of telephone conversations over a much smaller 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 variables. The compression variable sequence is further processed 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 available serial digital communication channel. Numerical feedback 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 utilized to establish a "virtual circuit" between sender and receiver.
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