• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.10
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• pp.11-20
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• 2014

In this paper, a novel transmission scheme is proposed to improve the data rates of spatial modulation (SM) which has low complexity and improves the spectral efficiency in correlated channel environments. The conventional SM scheme utilizes partial multiple antennas to transmit signal constellations and additional bits using antenna combinations. Therefore the channel capacity of SM is less than that of the conventional multiple input-multiple output (MIMO) scheme which uses all the available antennas. In this paper, an SM transmission scheme is proposed to improve the channel capacity using a tight frame pre-coder. The improvement in channel capacity of the SM scheme will be shown using computer simulations.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1211-1214
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• 1998

Real-time implementation of Conjugate-Structure Algebraic CELP(CS-ACELP) is presented. ITU-T Study Group(SG) 15 has standardized the CS-ACELP speech coding algorithm as G.729. A real-time implementation of the CS-ACELP is achieved using 16 bit fixed point DSP16210 Digital Signal Processor (DSP) of Lucent Technologies. The speech coder has been implemented in the bit-exact manner using the fixed point CS-ACELP C source which is the part of the G.729 standard. To provide a multi-channel vocoder solution to digital communication system, we try to minimize the complexity(e.g., MIPS, ROM, RAM) of CS-ACELP. Our speech coder shows 15.5 MIPS in performance which enables 4 channel CS-ACELP to be processed with one DSP16210.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.16 no.4
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• pp.276-280
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• 2016

A fixed rate speech coder based on the filter bank and the non-uniform sampling technique is proposed. The non-uniform sampling is achieved by the detection of inflection points (IPs). A speech block is band passed by the filter bank, and the subband signals are processed by the IP detector, and the detected IP patterns are compared with entries of the IP database. For each subband signal, the address of the closest member of the database and the energy of the IP pattern are transmitted through channel. In the receiver, the decoder recovers the subband signals using the received addresses and the energy information, and reconstructs the speech via the filter bank summation. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is confirmed. The signal-to-noise ratio (SNR) performance of the proposed method is comparable to that of the uniform sampled pulse code modulation (PCM) below 20 kbps data rate.

• The Transactions of the Korean Institute of Electrical Engineers A
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• v.48 no.12
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• pp.1589-1598
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• 1999

Conventional MSE or PSNR based methods for objective picture quality assessment of moving picture coder are not well correlated with subjective human evaluation. In recent years, the design of better objective quality assessment has attracted much intention and several picture quality metrics based on the properties of Human Visual System has been proposed. This paper proposes new metric which is appropriate for objective picture quality assessment of block based moving picture coder by considering frequency sensitivity, inter-intra channel masking and several distortion artifacts caused by block based coding. The experimental results show that the proposed method is good correlated with subjective assessment.

• Journal of Broadcast Engineering
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• v.7 no.2
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• pp.192-201
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• 2002

This paper develops a novel robust information hiding technique that uses channel codes derived from the error-correcting coder. The message encoded by the cover encoder is hidden in DCT transform domain of the cover image. The method exploits the sensitivity of human eyes to adaptively embed a visually recognizable message in an image without affecting the perceptual quality of the underlying cover image. Experimental results show that the proposed data hiding technique is robust to cropping operations, lossy JPEG compression, noise interference and secure against known stego attacks. The performance of the proposed scheme with turbo coder is superior to that without turbo coder.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.31-40
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• 2005

In this paper, we develop an approach toward JSC(Joint Source-Channel Coding) method for MPEG-4 based FGS(Fine Granular Scalability) video coding and transmission in fixed and mobile receiving environment(Digital Audio Broadcasting, DAB). The source coder used MPEG-4 FGS video codec, the channel coder used RCPC(Rate Compatible Punctured Convolution) code and the modulation method used QPSK modulation. We have considered channel environment of AWGN and mobile receiving environment. This study determined optimum Trade-off point between source bit rate and channel coding rate in variable channel states. We compared FGS-JSC method and general single layer CBR(Constant Bit Rate) transmission. In this results, FGS-JSC was appeared better performance than CBR transmission.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1806-1814
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• 1993

Efficient speech coders using tree coding combined with harmonic scaling are designed at the rate of 4.8 kilobitts/sec (kbps). A time domain harmonic scaling algorithm (TDHS) is used to compress input speech by a factor of two. This process allows the tree coder have 1.5 bits/sample for 4.8 kbps in the case of a 6.4 kHz sampling rate. In the backward adaptive tree coder, there are three components of the code generator, including a hybrid adaptive quantizer, a short-term predictor and a pitch predictor. The robustness of the tree coder is achieved by carefully choosing the input of the short term predictor adaptation. Also, inclusion of a smoother in the pitch predictor improves the error performance of tree coder in the noisy channel. Subjectively, tree coding combined with TDHS provides good quality speech at 4.8 kbps.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.3
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• pp.348-354
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• 1993

In this research, a multi-tree coder at 9.6Kbits/sec using a novel scheme for adaptation of the short-term coefficients is developed. The overall delay of the tree coder is maintained at 2.5 msec(16 samples at the 6.4KHz sampling frequency). This coder produces good quality speech over ideal channels, and it is very robust to channel errors up to a bit error rate (BER) of $10^{-3}$. This robustness is achieved by using a parallel adaptation scheme in combination with the use of a smoothed version of the received excitation sequence for adaptation of the short-term prediction coefficients. For the multi-tree coder, reconstructed output speech is evaluated using signal-to-quantization noise ratios (SNR), segmental SNRs, and informal listening tests.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.1
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• pp.85-100
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• 1991

In the discussion of digital mobile communication systems the speech coder based on the high quality low bit rate is an essential part of topics to overcome the limited availability of radio spectrum, which will enhance the communication services. In this paper we present the implementation and performance evaluation of 13kbps RPE LTP speech coder. An implementation of a real time full duplex coder with 75% of DSP loading rate using a single DSP chip has been shown, and also the fixed point simulations for H/W implementation has been performed. Finally, analysis result for relative bit importance of each transmitting parameter has been shown for channel coding.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5C
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• pp.295-301
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• 2011

In [1], it has been shown that the energy spreading transform (EST) based iterative equalizer (IE) could enhance its performance by improving the reliability of the decision feedback symbols without the help of the complicated channel decoder. In the matched filter (MF) based IE proposed in [1], however, its feedforward filter (FFF) has been designed in the frequency domain while its feedback filter (FBF) in the time domain. So its complexity increases proportional to the channel memory length. To solve this problem, in this paper, both FFF and FBF are designed in the frequency domain. This enables the proposed frequency domain IE (FD-IE) to achieve the lower complexity over the conventional method in the highly dispersive channel. In addition, simulation results demonstrate that the BER performance of the proposed method is the same as the conventional.