摘要:
A method and apparatus for estimating channel impulse response and data in a signal transmitted over a channel in a communication system. The channel impulse response is estimated uses correlative channel sounding, and then, using the estimated channel impulse response, the data in the signal is estimated. The output is then fed back to the channel impulse response estimator and the channel impulse estimation is repeated. The data estimation and channel impulse response estimation may be iterated a number of times.
摘要:
The invention relates to a method for converting a signal data rate and to a transmitter in a digital data transmission system, the transmitter comprising a number of data sources that generate a signal having a first data rate, and a coder for forming the signal into frames of a given length, which length corresponds to a second data rate, and a coder for grouping the signal into a first block the length of which corresponds to the first data rate. In order to convert the data rate flexibly, the transmitter comprises a coder, which calculates the difference between the frame and the number of symbols in the first block by subtracting the frame length from the number of symbols in the first block, which equalizes the difference by removing or repeating every ith symbol in the first block, i being determined as the nearest bigger or an equal integer when the number of symbols in the first block is divided by the absolute value of the difference that was calculated, which updates the number of the symbols that do not fit in the frame by subtracting therefrom the number of the symbols that have been repeated, and which forms the first block to consist of the symbols that have not been repeated or removed.
摘要:
A method and apparatus are disclosed for improving channel equalization and level learning in a data communication system. The disclosed equalizer training process separately updates the feed forward filter (FFF) and the level adapter, to gain additional improvements in the training of the feed forward filter (FFF). The multi-step equalizer training process initially trains the feed forward filter (FFF) using a two-level signal EQTR(n) having an ideal value (Step One) to help converge the feed forward filter (FFF) to a certain level. Once the feed forward filter (FFF) reaches a certain level of convergence, the training circuitry is reconfigured during step two of the equalizer training process, to evaluate and update the actual level of the signal EQTR(n), to compensate for the channel. The determined weighting factors are applied to a low pass filter and the actual level of the signal EQTR(n), B(n), is calculated. Once the actual level of the signal EQTR(n), B(n), has been calculated, the actual level of the signal EQTR(n), B(n), is applied to the level adapter, and the level adapter is no longer updated by reconfiguring the training circuitry to remove the error signal, err(n), inputs to the level adapter. During step three, the feed forward filter (FFF) continues to be updated and fine-tuned by the error signal err(n). Since the level of the signal EQTR(n) is the actual value, B(n), the performance of the feed forward filter (FFF) is improved. Once the equalizer training process is complete, the feed forward filter (FFF) is fixed. The improved training of the feed forward filter (FFF) allows the structure of level learning process to be simplified, with the training circuitry removed and the feed forward filter (FFF) fixed, where each level will be divided into six phases and processed individually.
摘要:
An automatic equalizer has a sampling-clock producing arrangement which is for, before selecting a sample timing, producing a sampling clock at the rate of L times of that after selecting, and after selecting the sample timing, producing a tap-coefficient selection signal according to the sample timing, and a sampling clock at the rate of 1/L times of that before selecting, according to the sample timing. In the sampling-clock producing arrangement, demodulation components are obtained in absolute values of impulse-response signals with respect to L sample timings, respectively. A selecting arrangement selects the sample timing by the use of the demodulation components. The impulse-response signals are produced in response to a sampled received-signal obtained by sampling a received signal with the sampling clock.
摘要:
A system and method for establishing an integrated forward error correction (FEC) scheme to perform multi-rate encoding on different priority data bits of a channel access message transmitted on a random access channel between devices of a communications network, such as between an access terminal and a base station of a satellite-based communications network. The channel access message includes a first data group representing first information and a second data group representing second information, which is transmitted between an access terminal and a base station in a satellite-based communications network. The system and method encodes the second data group at an encoding rate to provide a second encoded data group, and encodes the first data group at the same encoding rate to provide a first encoded data group. The encoding of the first and second data groups is performed by a single encoder, such as a rate ¼ convolutional encoder. The second encoded data group is transmitted from the access terminal to the base over a random access channel. The second encoded data group further can be punctured during transmission to in effect decrease its coding rate, for example, to rate ½ coding. The first encoded data group is transmitted from the access terminal to the base station, and is then retransmitted from the access terminal to the base station to in effect increase the rate of coding of the first encoded data group to, for example, ⅛ coding. At the base station, a combiner/demodulator combines the transmitted and retransmitted first encoded data group, and the combined first encoded data groups and the second encoded data group are then decoded by a decoder.
摘要:
A distortion compensation filtering mechanism for a direct spread-spectrum radio receiver comprises an iteratively adaptive FIR filter installed in the received signal processing path of the radio just upstream of the despreading function. The filter may be implemented as a relatively small numbered tap filter, having its precursor tap fixed at a maximum value. The remaining filter tap values are individually adaptively adjusted by the radio's control processor, which executes a tap adjustment routine to iteratively increment or decrement each variable tap value of the FIR filter to an ‘optimized’ value, that effectively minimizes the total (I and Q) power in the error in the data decisions performed by data signal analyzer.
摘要:
A finite impulse response (FIR) filter for wave-shaping digital quadrature amplitude modulation (QAM) symbols is disclosed, in which multipliers are replaced with multiplexers, the replaced multiplexers are utilized to receive the symbols directly from a symbol encoder without zero (0) interpolations, and the critical path is reduced by shifting the position of a delay device. The filter includes a first FIR means for delaying the externally inputted symbol data, and for utilizing the delayed symbol data as selection signals to sum up the selected multiplication product (selected from among products obtained by multiplying the symbol values by a pre-set filter tab coefficient) and the selected value selected by a first multiplexing means. A second FIR means delays again the delayed symbol data of the first FIR means, and utilizes the delayed symbol data as selection signals to sum up the selected multiplication product and the output value of the first FIR means.
摘要:
A method and apparatus for frequency domain equalization. The method and apparatus are particularly well suited for use in a receiver in a multicarrier transmission system that has predetermined symbols periodically embedded in the transmissions, such as in an ADSL system. The equalizer apparatus includes a digital filter having a plurality of single-tap filters. The filters operate on the symbols in the frequency domain, accepting frequency domain representations of the received symbols. The equalizer also includes a reference symbol generator that provides reference symbols, a coefficient generator that accepts the equalized frequency domain symbols from the digital filter and the reference symbols and updates the filter taps using a received predetermined symbol and the reference symbol. The coefficient generator includes a generator filter having a first and a second adaptation increment, an error generator, and a threshold detector that controls the coefficient generator in response to said error signal. Preferably, the threshold detector determines whether the equalizer output would have been decoded improperly during a synchronization symbol, and if so, enables the use of the second update increment in the generator filter.
摘要:
Digital signals provided by a repeater connected to a plurality of clients by unshielded twisted wire pairs, are converted to analog signals which become degraded during transmission through the wires. Clients convert the degraded analog signals to digital signals. Digital signal phases are coarsely adjusted to have assumed zero crossing times coincide in-time with a clock signal zero crossing. Signal polarity, and the polarity of any change, is determined at the assumed zero crossing times of the digital signals. Pre-cursor and post-cursor responses, resulting from signal degradation, are respectively inhibited by a feed forward and a decision feedback equalizer. The time duration of post-cursor response is further inhibited by a high pass filter and a tail canceller. Phase adjustments are made, after response inhibition, by determining the polarity, and the polarity of any change, at the assumed zero crossing times. Before phase adjustments are made, a phase offset is provided in order to compensate for phase degradations introduced by the unshielded twisted wire pairs.
摘要:
A receiver has a dual mode of operation—a carrierless amplitude modulation/phase modulation (CAP) mode and a quadrature amplitude modulation (QAM) mode-yet only requires a single equalizer structure for both the CAP mode of operation and the QAM mode of operation during blind start-up. The receiver uses the same blind equalization updating algorithm independent of the type of received signal for converging the equalizer structure. The blind equalization updating algorithm incorporates a constant R, whose value is a function of the type of received signal, e.g., a QAM signal or a CAP signal. The type of received signal is determined as a function of the in-phase component of the mean-squared error, E[e2n.