摘要:
Frequency domain based methods and systems to perform adaptive multi-mode pre-decoding linear equalization, adaptive channel estimation, turbo equalization, turbo equalization iteration control, and noise variance estimation. An adaptive linear pre-decoder equalizer may include a multi-mode equalizer, which may include a blind equalizer to determine initial equalization coefficients and a decision-directed equalizer to refine the initial equalization coefficients. Turbo equalization may include a dynamic iteration termination criterion and/or a fixed termination criterion. Turbo equalization may be based in part on an adaptive estimated noise variance. Methods and systems disclosed herein may be implemented with respect to single-carrier signals such as digital video broadcast signals.
摘要:
A method and circuit for performing channel equalization in a high speed transmission system comprising a transmitter and receiver. An application specific digital signal processor, ASDSP, performs channel equalization and compensation on digital data received from an analogue-to digital converter of the receiver. The ASDSP is operable to execute an application specific set of op-codes needed for performing channel equalization and compensation. An ASDSP register is coupled between the ASDSP and a system CPU in a feedback loop for performing channel equalization at the receiver. The ASDSP stores equalizer parameters and bit error rate measurements used by the ASDSP for performing channel equalization and compensation. An ASDSP program storage memory, coupled to and accessible by the ASDSP, stores an ASDSP micro-sequence program for controlling the processing steps for channel equalization and dataflow through the ASDSP.
摘要:
A method and apparatus are provided for reducing a channel length. The method implements a channel-length reduction filter that takes into account a pulse response of the channel. The method includes the following steps: cutting the filter into at least one first and one second portion; optimizing said first portion of the filter according to a first criterion in order to output a first set of filtering coefficients; reducing the length of the channel by optimizing said second portion of the filter according to a second criterion different from the first one and based on the first set of filtering coefficients of said first portion of the filter in order to output a second set of filtering coefficients.
摘要:
The problem of inefficient channel impulse-response processing is addressed by processing different parts of a channel impulse response to accurately locate channel taps, and to generate more than one set of equalization coefficients. This allows the most-suited equalization coefficient to be selected based on a selection criterion.
摘要:
Methods and apparatus to facilitate improve code division multiple access (CDMA) receivers are disclosed. An example method disclosed herein comprises: receiving a signal containing first portions that are based on known data and second portions that are based on unknown data; generating a training signal, from the received signal, that substantially represents one or more of the first portions; adapting filter coefficients using the training signal; and equalizing the received signal using the adapted filter coefficients.
摘要:
Provided are an apparatus and method for selecting an optimal signal using auxiliary equalization in a diversity receiver. The optimal signal selecting apparatus includes: a plurality of sync recovery units for extracting sync information from baseband signals, which are candidate signals, except a baseband signal selected as a current optimal signal a plurality of auxiliary equalizers for channel-equalizing the candidate signals based on the extracted sync information; a plurality of SNR measuring units for measuring signal-to-noise ratios (SNRs) of the candidate signals inputted to the auxiliary equalizers and the candidates signals equalized in the auxiliary equalizers; and an optimal signal selector for selecting an optimal candidate signal from the candidate signals by using the extracted sync information and the measured SNRs, and replacing the optimal signal with the optimal candidate signal when reception quality of the current optimal signal is poor.
摘要:
An adaptive, reduced-complexity soft-output maximum-likelihood detector that is operable to process data by adaptively selecting a processing scheme based on a determination of signal quality. The signal quality is derived as a function of the noise, the modulation format, the channel (the communication environment), the transmit signal power and the receive signal power. If the signal quality is low, the signal is processed using a maximum likelihood detector. If, however, the signal quality is high, a simpler sub-optimal detector is used. By estimating the signal quality and choosing an appropriate detection method, the present invention ensures accurate detection of incoming data signals in a MIMO communication system while maintaining the highest possible processing speed.
摘要:
Embodiments of the present invention provide a method, apparatus and system for detecting a transmitted signal according to a detection algorithm selected from two or more detection algorithms based on a predetermined selection criterion.
摘要:
Aspects of a method and system for decoding single antenna interference cancellation (SAIC) and redundancy processing adaptation using frame process are provided. A receiver may decode video, voice, and/or speech bit sequences based on a first decoding algorithm that may utilize data redundancy and that may impose physical constraints. The receiver may also decode a bit sequence based on a second decoding algorithm that utilizes SAIC. The first and second decoding algorithms may be adapted to perform in parallel and a decoded received bit sequence may be selected based on a redundancy verification parameter. The first and second decoding algorithms may also be adapted to be performed sequentially where the subsequent decoding operation may be conditioned to the initial decoding operation. Moreover, either the first or the second decoding algorithm may be selected for decoding the received bit sequence. The selection may be based on noise and/or interference measurements.
摘要:
A number of basic equalization and demodulation structures have been shown to be appropriate for DMT systems depending on the channel, noise, and system parameters. These include single path, dual path, oversampled, and double rate structures. Using the fundamental computation units of two TEQs (FIR filters) and two FFTs, in conjunction with simple delays, downsampling and routing, single path, dual path, oversampled and double rate equalization structures can be realized from a common equalization structure.