Title :
A filterbank structure for voice-band PCM channel pre-equalization
Author :
Alagha, Nader Sheikholeslami ; Kabal, Peter
Author_Institution :
Dept. of Electr. & Comput. Eng., McGill Univ., Montreal, Que., Canada
Abstract :
A non-maximally decimated filterbank structure for pre-equalizing channels with intersymbol interference (ISI) is investigated. The impulse response of the channel is assumed to be known at the transmitter. Compared with the classical Tomlinson-Harashima (1971) precoding technique, the proposed pre-equalizer compensates for the channel without increasing the number of the received signal levels (channel alphabet). The proposed technique does not require the channel to be minimum-phase. The filterbank structure adds redundancy to the input signal to compensate for the channel ISI while keeping the transmitted power bounded. The proposed pre-equalization is particularly useful for data transmission over voice-band PCM channels. The upstream PCM channel is bandlimited, causing severe ISI at the output of the front-end receiver filter. By using the pre-equalizer at the transmitter, channel ISI can be mitigated
Keywords :
FIR filters; bandlimited communication; channel bank filters; data communication; encoding; equalisers; intersymbol interference; pulse code modulation; signal sampling; subscriber loops; telecommunication channels; voice communication; FIR filter; ISI; PSTN; Tomlinson-Harashima precoding technique; bandlimited upstream PCM channel; bounded transmitted power; channel alphabet; channel compensation; channel impulse response; data transmission; down-sampling; front-end receiver filter; input signal redundancy; interference suppression; intersymbol interference; non-maximally decimated filterbank structure; public switching telephone network; received signal levels; subscriber loop; time-varying precoder; transmitter; up-sampling; voice-band PCM channel pre-equalization; Bandwidth; Central office; Filter bank; Modems; Phase change materials; Pulse modulation; Quantization; Sampling methods; Subscriber loops; Telephony;
Conference_Titel :
Acoustics, Speech, and Signal Processing, 2000. ICASSP '00. Proceedings. 2000 IEEE International Conference on
Conference_Location :
Istanbul
Print_ISBN :
0-7803-6293-4
DOI :
10.1109/ICASSP.2000.861086