Multi-carrier code division multiple access (MC-CDMA) has been considered as a strong candidate for next generation wireless communication system due to its excellent performance in multi-path fading channel and simpl...
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ISBN:
(纸本)9781424456383
Multi-carrier code division multiple access (MC-CDMA) has been considered as a strong candidate for next generation wireless communication system due to its excellent performance in multi-path fading channel and simple receiver structure. However, like all the multi-carrier transmission technologies such as OFDM, the inter-carrier interference (ICI) produced by the frequency offset between the transmitter and receiver local oscillators or by Doppler shift due to high mobility causes significant BER (bit error rate) performance degradation in MC-CDMA system. Many ICI cancellation methods such as windowing and frequency domain coding have been proposed in the literature to cancel ICI and improve the BER performance for multi-carrier transmission technologies. However, existing ICI cancellation methods do not cancel ICI entirely and the BER performance after ICI cancellation is still much worse than the BER performance of original system without ICI. Moreover, popular ICI cancellation methods like ICI self-cancellation reduce ICI at the price of lowering the transmission rate and reducing the bandwidth efficiency. Other frequency-domaincoding methods do not reduce the data rate, but produce less reduction in ICI as well. In this paper, we propose a novel ICI cancellation scheme that can eliminate the ICI entirely and offer a MC-CDMA mobile system with the same BER performance of a MC-CDMA system without ICI. More importantly, the proposed ICI cancellation scheme (namely Total ICI Cancellation) does not lower the transmission rate or reduce the bandwidth efficiency. Specifically, by exploiting frequency offset quantization, the proposed scheme takes advantage of the orthogonality of the ICI matrix and offers perfect ICI cancellation and significant BER improvement at linearly growing cost. Simulation results in AWGN channel and multi-path fading channel confirm the excellent performance of the proposed Total ICI Cancellation scheme in the presence of frequency offset o
Adaptive transform coding incorporating Time domain Aliasing Cancellation (TDAC) is presented in this paper for speech coding at low data rates. The TDAC transform differs from the conventional Discrete Cosine Transfo...
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Adaptive transform coding incorporating Time domain Aliasing Cancellation (TDAC) is presented in this paper for speech coding at low data rates. The TDAC transform differs from the conventional Discrete Cosine Transform (DCT) in that the analysis/synthesis window length can be longer than the system decimation factor. The channel bandwidth of the TDAC transform is narrower when compared to the DCT and therefore results in less spectral interaction occurring between bands. The TDAC transform coder employs forward adaptive scalar quantization of the transform coefficients and dynamic bit allocation over a block time corresponding to three transform operations. The associated side information is vector quantized, resulting in a side information overhead of 1.625 K bits per second. Both subjective and objective comparisons were made between TDAC and DCT coders at data rates of 9.6 K, 12 K and 16 K bits per second. The objective comparison involved segmental signal to noise ratio (SNRSEG) measurements while the subjective quality assessment was determined from A-B comparisons of recovered speech passed through the TDAC coder, the DCT coder and log PCM coders. The results indicate the TDAC transform coder is both objectively and subjectively superior to a transform coder employing the conventional DCT. Dieser Beitrag beschäftigt sich mit adaptiver Transformationscodierung von Sprachsignalen bei niedrigen Datenraten unter Verwendung des Verfahrens der Kompensation von Aliasverzerrungen im Zeitbereich (Time domain Aliasing Cancellation, TDAC). Die TDAC Transformation unterscheidet sich von der üblicherweise verwendeten diskreten Kosinustransformation (Discrete Cosine Transform, DCT) dadurch, daβ die Länge des für die Analyse und Synthese verwendeten Fensters gröβer sein kann als durch den Reduktionsfaktor des Systems vorgegeben. Die benötigte Bandbreite ist bei der TDAC Transformation kleiner als bei der DCT; hierdurch verringert sich die gegenseitige Beeinflussung benachba
Unified speech and audio codecs often use a frequency domain coding technique of the transform coded excitation (TCX) type. It is based on modeling the speech source with a linear predictor, spectral weighting by a pe...
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Unified speech and audio codecs often use a frequency domain coding technique of the transform coded excitation (TCX) type. It is based on modeling the speech source with a linear predictor, spectral weighting by a pe...
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ISBN:
(纸本)9781467369985
Unified speech and audio codecs often use a frequency domain coding technique of the transform coded excitation (TCX) type. It is based on modeling the speech source with a linear predictor, spectral weighting by a perceptual model and entropy coding of the frequency components. While previous approaches have used neighbouring frequency components to form a probability model for the entropy coder of spectral components, we propose to use the magnitude of the linear predictor to estimate the variance of spectral components. Since the linear predictor is transmitted in any case, this method does not require any additional side info. Subjective measurements show that the proposed methods give a statistically significant improvement in perceptual quality when the bit-rate is held constant. Consequently, the proposed method has been adopted to the 3GPP Enhanced Voice Services speech coding standard.
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