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 Developing a channel impulse response by using distortion

Details
Inventors: Wang, Jin-Der;
Assignee: Lucent Technologies Inc. (Murray Hill, NJ)
Primary Examiner: Ghebretinsae; Temesghen
Assistant Examiner:
Attorney, Agent or Firm: Opalach; Joseph J.

A digital audio broadcasting (DAB) system includes a radio-frequency (RF) receiver that develops a characterization signal representative of a respective mobile communications channel as a precursor to estimating a channel impulse response. The latter is used to set coefficients of an equalizer of the receiver to compensate for distortion such as intersymbol interference. In generating the channel impulse response a Fast Fourier Transform (FFT) of the characterization signal is performed. The resulting FFT is then clipped such that frequency components above a predetermined upper threshold and less than a predetermined lower threshold are set to the respective threshold values. This allows coefficients for the equalizer to be generated such that the equalizer converges even in the presence of strong reflections in a mobile communications environment.

DETAILED DESCRIPTION FIG.
1 shows a high-level block diagram of a DAB communications system 10 embodying the principles of the invention.
DAB communications system 10 comprises transmitter 100, communications channel 200, and receiver 300.
Before describing the details of the inventive concept, a general overview of the operation of DAB communications system 10 will be given.
Also, perceptual audio coding is well-known and will not be described in detail.
For example, see U.
S.
Pat.
No.
5,285,498, entitled "Method and Apparatus for Coding Audio Signals Based on Perceptual Model," issued Feb.
8, 1994 to Johnston.
Other such coding techniques are described, e.
g.
, in J.
P.
Princen and A.
B.
Bradley, "Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing Cancellation," IEEE Trans.
ASSP, Vol.
34, No.
5, October, 1986; E.
F.
Schroder and J.
J.
Platte, "`MSC`: Stereo Audio coding with CD-Quality and 256 kBIT/SEC," IEEE Trans.
on Consumer Electronics, Vol.
CE-33, No.
4, November 1987; Johnston, "Transform Coding of Audio Signals Using Noise Criteria," IEEE J.
S.
C.
A.
, Vol.
6, No.
2, February 1988; and U.
S.
Pat.
No.
5,341,457, entitled "Perceptual Coding of Audio Signals," issued Aug.
23, 1994 to Hall et al.
In FIG.
1, an analog audio signal 101 is fed into preprocessor 105 where it is sampled (typically at 48 Khz) and converted into a digital pulse code modulation (PCM) signal 106 (typically 16 bits) in standard fashion.
The PCM signal 106 is fed into a perceptual audio coder (PAC) 110, which compresses the PCM signal and outputs compressed PAC signal 111.
The latter represents a 170 kb/s bit stream of which 10 kb/s represents a control channel for ancillary data and 160 kb/s represents the compressed audio signal.
Compressed PAC signal 111 is applied to error protection coder 115, which applies a Reed-Solomon code to provide 100% redundancy to compressed PAC signal 111.
It is assumed that error protection coder 115 also includes a buffer and interleaver (not shown) to further combat the effects of communications channel 200



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