1-3hit |
Feng YANG WenJun ZHANG ShuRong JIAO Xiaoyun HOU
Intercarrier interference will cause the loss of subchannel orthogonality and increase the error floor in proportion to the Doppler frequency. In this paper, we firstly analyze the generation mechanism of intercarrier interference in OFDM. Then we propose an O(N log2N) complexity ICI equalizer for OFDM systems in the presence of double selective fading which is mainly bases on FFT operation. Simulation result shows that with only 6 iterations LCD-FFT can achieve better performance than the LS-equalizer. After 10 iterations LCD-FFT performs almost the same as MMSE equalizer.
Chengyu LIN Wenjun ZHANG Feng YANG Youyun XU
To improve the performance of the optimal pilot sequences over multiple OFDM symbols in fast time-varying channels, this letter proposes a novel channel estimation method using virtual pilot tones in multiple-input multiple-output (MIMO) orthogonal frequency division multiplexing (OFDM) systems. Assuming that the superimposed virtual pilot tones at the data locations over the specific sub-carriers are transmitted from all transmit antennas, the corresponding virtual received pilot signals at the same locations are obtained from the neighboring real received pilot signals over the same sub-carriers by Wiener filter. Based on the least squares (LS) channel estimation, the channel parameters can be obtained from the combination of the virtual and real received pilot signals over one OFDM symbol. Simulation results show that the proposed channel estimation method greatly outperforms the previous method for the optimal pilot sequences over multiple OFDM symbols in fast time-varying channels, as well as approaches the method for the comb-type optimal pilot sequences in performance.
Jianguo TAN Wenjun ZHANG Peilin LIU
Sinusoidal representation has been widely applied to speech modification, low bit rate speech and audio coding. Usually, speech signal is analyzed and synthesized using the overlap-add algorithm or the peak-picking algorithm. But the overlap-add algorithm is well known for high computational complexity and the peak-picking algorithm cannot track the transient and syllabic variation well. In this letter, both algorithms are applied to speech analysis/synthesis. Peaks are picked in the curve of power spectral density for speech signal; the frequencies corresponding to these peaks are arranged according to the descending orders of their corresponding power spectral densities. These frequencies are regarded as the candidate frequencies to determine the corresponding amplitudes and initial phases according to the least mean square error criterion. The summation of the extracted sinusoidal components is used to successively approach the original speech signal. The results show that the proposed algorithm can track the transient and syllabic variation and can attain the good synthesized speech signal with low computational complexity.