摘要:
A method of selecting a PCM codeword set. A communication system includes an encoder and a decoder. The encoder has a digital connection to the digital portion of the telephone network. The digital portion of the telephone network is connected by an analog loop to the decoder. The method includes transmitting sequence of PCM codewords from the encoder and receiving the sequence as a sequence of analog voltages. A noise level it determined and a minimum separation value is selected in response to the noise level. A set of PCM codewords is selected such that no two PCM codewords within the set are separated by less than minimum separation value.
摘要:
The present invention relates to a concatenated convolutional encoder and decoder for the next generation mobile communication system requiring a high performance channel coding, in particular to a concatenated convolutional encoder and decoder of a mobile communication system which is capable of providing a dual mode encoder and decoder for supporting both a parallel concatenated convolutional code and a serially concatenated convolutional code and improving the performance of the system by using punctured and thrown away sequence in a convolutional encoder. The present invention can show stable performance regardless of SNR, accordingly the credibility of the system can increase.
摘要:
A first transceiver transmits a set of test levels to a second transceiver through a communication channel with one or more types of companding laws. The second transceiver determines line encoding with, and conversion between, the companding laws present in the communication channel based on the received set of test signals. The set of test levels are signals having levels determined based on the difference between the normalized amplitude, vertex, or energy curves for the types of companding laws, with or without accounting for other sources of network distortion. Additional distortion from line characteristics, such as robbed-bit signaling (RBS) and/or line impairment, may be detected based on changes in encoding sample levels of transmitted test signals that are reconstructed by the second transceiver. The second transceiver may then transmit information to the first transceiver about the companding laws and other sources of distortion present in the network. The second transceiver employs a method of constellation adjustment to correct for distortion resulting from line encoding, encoding conversion, RBS and other line impairments. For a given, detected encoding conversion during the training phase, two constellations are employed, one for the first transceiver and one for the second transceiver. For constellation adjustment, each transceiver first detects encoding, encoding conversion, RBS, and other line impairments using a set of PCM test levels during their respective training phases. The second transceiver then adjusts its transmit constellation for communication with the first transceiver based on the detected encoding, encoding conversion, RBS, and other line impairments.
摘要:
A method of encoding information of an input signal using a fixed number of bits for each unit time frame. Part of the encoded information of at least one second frame temporally consecutively or non-consecutively preceding or following a first frame is contained in the encoded information of the first frame. This eliminates fluctuations in the sound quality due to bit surplus/shortage resulting from quantization for achieving efficient encoding and decoding.
摘要:
Techniques to determine data rates for a number of data streams transmitted via a number of transmission channels (or transmit antennas) in a multi-channel (e.g., MIMO) communication system. In one method, the “required” SNR for each data rate to be used is initially determined, with at least two data rates being unequal. The “effective” SNR for each data stream is also determined based on the received SNR and successive interference cancellation processing at the receiver to recover the data streams. The required SNR for each data stream is then compared against its effective SNR. The data rates are deemed to be supported if the required SNR for each data stream is less than or equal to its effective SNR. A number of sets of data rates may be evaluated, and the rate set associated with the minimum received SNR may be selected for use for the data streams.
摘要:
An asynchronous digital system, an asynchronous data path circuit, an asynchronous digital signal processing circuit and an asynchronous digital signal processing method, which enables improved processing speed while maintaining high reliability are provided by dividing the overall chip into blocks with a specified area, forming the connection between the blocks by applying thereto a delay insensitive (Dl) model or a quasi delay insensitive (QDI) model, while forming each block by applying thereto a scalable delay insensitive (SDI) model. In the SDI model, the system is configured using circuit components having a delay assumed during design in which if the specification states that a signal transition (b) in a subcircuit 7 precedes a signal transition (c) in a subcircuit 8, &kgr;·Tab
摘要:
A method is described for convolutionally encoding and decoding data (voice coded data), organized into (35-bit, 20 msec) frames, where the data is encoded by a finite-state data encoder (RSC coder 216) for transmission over a data channel (18). The RSC encoder avoids the need for termination bits associated with each N-bit frame of data, so that the number of bits associated with each encoded frame is reduced, and the throughput of the channel can be increased. The method according to the invention includes storing the first M bits of each frame. Once they are stored, they are loaded in parallel into the M stages of the encoder, thereby deleting residual states from the preceding frame. The remaining (N−M) bits are then applied to the encoder, causing it to produce the convolutional code. After the last of the (N−M) bits are encoded, the M stored bits are summed with feedback from the encoder, and reapplied to the encoder. The ending state of the encoder is thus equal to the starting state.
摘要:
A block sliding window data decoder includes a forward recursion calculator and a plurality of backward recursion calculators including a first backward recursion calculator and a second backward recursion calculator that receives inputs from the first backward recursion calculator. The first backward recursion calculator operates every D cycles to perform a backward recursion over received input signals, while assuming that all future states are equally likely. The decoder further includes a symbol probability calculator that receives inputs from the forward recursion calculator and from the first backward recursion calculator. A memory of the decoder stores the input signals and is organized as N cells, wherein for each cycle one of the N cells is written while N−1 of the cells are read and their contents provided to the forward recursion calculator and to the first and second backward recursion calculators. Each calculation cell of the decoder includes a normalizer for normalizing the input signal and signals processed by the forward recursion calculator and the first and second backward recursion calculators. The normalizer is implemented using AND functions.
摘要:
A method of transmitting a quite, or zero, signal in a PCM communication system. The zero signal is specified universally in ordered set terms for either &mgr;-Law or A-Law PCM systems, which has minimal energy in the A-Law system, and contains no energy within the &mgr;-Law system. The signal is preferably specified as a repetition of six intervals (or multiple thereof) because the DTN can modify PCM codes on a six interval period by the robbed-bit signaling mechanism. The zero signal may be used to detect network elements that produce single-signed zero outputs from zero inputs of either sign. In addition, by examination of the zero signal the receiver may determine whether the channel includes an analog link or connection.
摘要:
A method of determining digital channel attenuation; comprising the steps of: receiving a known training sequence of PCM codes, which PCM codes are subjected to the attenuation within the digital channel; quantizing the received known training sequence of PCM codes according to a predetermined thresholding procedure; identifying identical PCM codes created as a result of the thresholding procedure; and, determining the attenuation of the digital channel based upon the identification of identical PCM codes. A method is also disclosed for determining a digital channel PCM code transformation comprising receiving a known training sequence of PCM codes, which PCM codes are subjected to the PCM code transformation within the digital channel, quantizing the received known training sequence of PCM codes according to a predetermined thresholding procedure, and determining the transformation of transmitted codes to those received. A method is also disclosed for improved echo cancellation in a communications network having an analog and a digital modem, comprising saving codes transmitted from the digital modem to the analog modem for echo cancellation, transforming, by a mapping table, codes transmitted from said digital modem to codes received by the analog modem, and, using the received codes as a reference signal for cancellation of echo. A method of improved spectral shaping using a transmit shaping transfer function in a communications network having an analog and a digital modem, comprising, transforming, by a mapping table, codes transmitted from the digital modem to codes received by the analog modem, using the received codes for transformation to their linear value equivalent representations, and, applying the linear value representations to the transmit shaping transfer function.