摘要:
A fast Fourier transformation is performed on a first vector of signals, and as a result a second vector of signals is provided. A feed forward equalization is performed by multiplying each of the components of said second vector with equalization parameters, and as a result a third vector of signals is provided. An inverse fast Fourier transformation is performed on said third vector, and as a result a fourth vector of signals is provided. An output signal of said first section is provided on the basis of said fourth vector of signals. In a second section a signal derived from an output signal of said second section is is filtered via linear feedback filtering and the filtered signal is added to said first section output signal, and an added signal is provided, and said second section output signal is generated by extracting samples from said added signal.
摘要:
Disclosed is a frequency-domain decision feedback equalizing method and device for single carrier modulation, preferably for use in a broadband communication system, wherein in a first section a fast Fourier transformation is performed on a first vector of signals inputted, and as a result a second vector of signals is provided, a feed forward equalization is performed by multiplying each of the components of said second vector of signals with equalization parameters, and as a result a third vector of signals is provided, an inverse fast Fourier transformation is performed on said third vector of signals, and as a result a fourth vector of signals is provided, and an output signal of said first section is provided on the basis of said fourth vector of signals; and in a second section a linear feedback filtering of a signal derived from an output signal of said second section is performed, and a filtered signal is provided, said filtered signal is added to said output signal of said first section, and an added signal is provided, and said output signal of said second section is generated by extracting samples from said added signal.
摘要:
A wireless CDMA communication system receiver receives a stream of chips generated by spreading data symbols formed by grouping bits of information at a wireless CDMA communication transmitter which are broadcast at a certain chip-rate. The received chips are de-spread and symbols pertaining to respective users are reconstructed. The stream of chips are formatted into blocks of chips, and an iterative block decision feedback equalization is performed in a frequency domain at the chip-rate of the broadcast stream of chips to remove inter-symbol interference by defining a transfer function. The chips generated are interleaved by spreading each data symbol being transmitted before broadcasting the stream of interleaved chips in distinct blocks of chips.
摘要:
A method for canceling interference at a wireless code division multiple access (CDMA) communication receiver is provided. The wireless CDMA communication system receiver receives a stream of chips generated by spreading data symbols formed by grouping bits of information at a wireless CDMA communication transmitter which are broadcast at a certain chip-rate. The received chips are de-spread and symbols pertaining to respective users are reconstructed. The method includes formatting the stream of chips into blocks of chips, and performing an iterative block decision feedback equalization in a frequency domain at the chip-rate of the broadcast stream of chips to remove inter-symbol interference by defining a transfer function. The transfer function is defined based upon iteration cycles as a function of data detected in a preceding iteration cycle. The chips generated are interleaved by spreading each data symbol being transmitted before broadcasting the stream of interleaved chips in distinct blocks of chips.
摘要:
Filterbank-based modulation systems comprise sender-processors (20,30) with inverse-fast-fourier-transformating-modules (23,33) and filtering-modules (24,34) and comprise receiver-processors (40) with fast-fourier-transformating-modules (43). Interference caused by said filtering-modules (24,34) is reduced by, in said sender-processors (20,30), introducing coding-modules (22,32) with further-filtering-modules (26,36) in feedback loops, and by, in said receiver-processors (40), introducing decoding-modules (44). Splitting-modules (21,31,41) and combining-modules (25,35,45) allow the use of signal streams and parallel filterbanks. Coding-modules (22 resp. 32) comprise sub-coding-modules (22-1,22-2, . . . , 22-a or 32-1,32-2, . . . ,32-b), filtering-modules (24 resp. 34) comprise sub-filtering-modules (24-1,24-2, . . . ,24-a or 34-1,34-2, . . . ,34-b), further-filtering-modules (26 resp. 36) comprise sub-further-filtering-modules (26-1,26-2, . . . ,26-a or 36-1,36-2, . . . ,36-b), and decoding-modules (44) comprise sub-decoding-modules (44-1,44-2, . . . ,44-c), all per signal stream. The sub-further-filtering-modules either receive input signals from outputs of said inverse-fast-fourier-transformating-modules and supply output signals via fast-fourier-transformating-modules to inputs of said sub-coding-modules via adding/subtracting-modules for reducing interference per signal stream (or per subcarrier/subband), or receive input signals from outputs of said sub-coding-modules and supply output signal to inputs of said sub-coding-modules via adding/subtracting-modules for reducing interference per signal stream (or per subcarrier/subband) as well as between signal streams (or between subcarriers/subbands) and introducing so-called fractionally spaced filterbank-based modulation systems.
摘要:
An unknown voiceband digital data modem signal is classified as being generated by one of a plurality of possible digital data modem signal sources, e.g., CCITT V.29, CCITT V.32, CCITT V.33 or the like digital data modems. Classification is achieved by employing a blind, i.e., self-recovering, adaptive equalizer to remove effects of linear channel impairments and to generate a sequence of magnitude estimates at the symbol rate of the unknown voiceband digital data modem signal. The sequence of magnitude estimates is compared to predetermined representations of known possible voiceband digital data modem signals and the results of the comparison are used to identify the digital data modem signal source of the signal. In one example, the predetermined representations are templates of conditional density functions of magnitude estimates obtained from known voiceband digital data modem signals generated by corresponding digital data modem signal sources. The symbol rate detector detects bit rate and type of modulation.
摘要:
A signal is classified as one among a plurality of classifications by employing the autocorrelation of a complex low-pass version of the signal, i.e., the complex autocorrelation. The normalized magnitude of the complex autocorrelation obtained at a prescribed delay interval, i.e., "lag", is compared to predetermined threshold values to classify the signal as having one of a plurality of baud rates.
摘要:
In an ADPCM coder and decoder including a so-called locking-unlocking adaptation speed control, the adaptation speed is locked to a very slow, almost constant, speed of adaptation for voiceband data and partial band energy signals, i.e., tones and tone like signals, and is unlocked to achieve a fast speed of adaptation for speech. When a so-called partial band energy signal is being inputted, the adaptation speed is biased toward the unlocked state and when a transition occurs from a partial band energy signal to another such signal, the adaptation speed is set to the totally unlocked state and coefficients of an adaptive predictor are set to prescribed values. This is done in both the coder and decoder to minimize generation of impulse noise in the decoder output.
摘要:
A signal is classified as being one among a plurality of classifications by employing a prescribed relationship between absolute moments of a complex low-pass version of the signal. Specifically, the prescribed relationship is related to the second order moment of the magnitude of the complex low-pass version being normalized by the first order moment squared. This results in a so-called normalized variance which is compared to predetermined threshold values to classify the signal as having one of a plurality of modulation schemes, e.g., FSK, PSK or QAM.In another embodiment, a signal is classified as being speech or voiceband data. This is achieved by employing a phase relationship, i.e., the sign, of the autocorrelation of a complex low-pass version of the signal and the normalized variance. If the autocorrelation has a prescribed phase or the normalized variance is greater than a predetermined value the signal is speech, otherwise it is voiceband data.