摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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).
摘要:
The voice signal s(n) is filtered through a short-term predictive filter (13) tuned with PARCOR derived coefficients computed over a pre-emphasized s(n), said filter (13) providing a short-term residual r(n). Said r(n) signal is then processed through a first Cod-Excited/Long-Term Predicative coder providing first couples of table address and gain data (k1, gl)'s. An error signal r'(n) is then derived by subtracting coded/decoded data from uncoded data. Then said error signal is processed through a second Code-Excited/Long-Term Predictive coder providing second couples of data (k2, g2)'s. Full rate coding is achieved by multiplexing both couples (k1, gl)'s and (k2, g2)'s into a multi-rate frame; while switching to a lower rate is achieved through a mere delation of (g2, k2)'s from the full rate frame.
摘要:
This low bit rate voice encoding involves short-term predictive filtering the voice signal s(n) using partial correlation related coefficients derived from pre-emphasized s(n), and deriving a short term signal r(n); then deriving a long-term residual signal e(n) by subtracting a delayed synthesized short term b.r'(n-31 M) from said r(n); and code excited encoding e(n) into codeword references k's and associated gains G's.
摘要:
This process enables cancelling residual echo signals in a transmission system wherein echo signals are partially cancelled through use of a filter generating an echo replica, and substracting said replica from the signal in the echo path. The process involves measuring energies of incoming and outgoing signals, comparing the ratio of said energies to a predetermined threshold value to generate a flag information, which flag is used to control a switch on the echo path. The threshold is determined, for instance by using the information provided by the echo path impulse response not considered for the generation of the echo replica.
摘要:
In a TASI system using dynamic subband allocation and BCPCM (Block Companded PCM), losses due to freeze-out or delay are avoided by use of a multi-rate macro-frame format, wherein switching to a lower rate (24, 16, or 8 Kbs) for one or more of the input channels is accommodated by simply dropping one or more sections of each signal frame, since the lower rate bits are distributed over the entire frame.