• Title/Summary/Keyword: Symbol Error Rate

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Study on CGM-LMS Hybrid Based Adaptive Beam Forming Algorithm for CDMA Uplink Channel (CDMA 상향채널용 CGM-LMS 접목 적응빔형성 알고리듬에 관한 연구)

  • Hong, Young-Jin
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.9C
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    • pp.895-904
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    • 2007
  • This paper proposes a robust sub-optimal smart antenna in Code Division Multiple Access (CDMA) basestation. It makes use of the property of the Least Mean Square (LMS) algorithm and the Conjugate Gradient Method (CGM) algorithm for beamforming processes. The weight update takes place at symbol level which follows the PN correlators of receiver module under the assumption that the post correlation desired signal power is far larger than the power of each of the interfering signals. The proposed algorithm is simple and has as low computational load as five times of the number of antenna elements(O(5N)) as a whole per each snapshot. The output Signal to Interference plus Noise Ratio (SINR) of the proposed smart antenna system when the weight vector reaches the steady state has been examined. It has been observed in computer simulations that proposed beamforming algorithm improves the SINR significantly compared to the single antenna case. The convergence property of the weight vector has also been investigated to show that the proposed hybrid algorithm performs better than CGM and LMS during the initial stage of the weight update iteration. The Bit Error Rate (BER) characteristics of the proposed array has also been shown as the processor input Signal to Noise Ratio (SNR) varies.

A Relay Selection Scheme with Q-Learning (Q-Learning을 이용한 릴레이 선택 기법)

  • Jung, Hong-Kyu;Kim, Kwang-Yul;Shin, Yo-An
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.49 no.6
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    • pp.39-47
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    • 2012
  • As a scheme to efficiently reduce the effects of multipath fading in next generation wireless communication systems, cooperative communication systems have recently come into the spotlight. Since these cooperative communication systems use cooperative relays with diverse fading coefficients to transmit information, having all relays participate in cooperative communication may result in unnecessary waste of resources, and thus relay selection schemes are required to efficiently use wireless resources. In this paper, we propose an efficient relay selection scheme through self-learning in cooperative wireless networks using Q-learning algorithm. In this scheme, we define states, actions and two rewards to achieve good SER (Symbol Error Rate) performance, while selecting a small number of cooperative relays. When these parameters are well-defined, we can obtain good performance. For demonstrating the superiority of the proposed Q-learning, We compared the proposed scheme with Q-learning and a relay selection scheme with a mathematical analysis. The simulation results show that, compared to a scheme that obtains optimum relays through a mathematical analysis, the proposed scheme uses resources efficiently by using smaller numbers of relays with comparable SER performance. According to these simulation results, the proposed scheme can be considered as a good attempt for future wireless communication.

The Performance Analysis of CCA Adaptive Equalization Algorithm for 16-QAM Signal (16-QAM 신호에 대한 CCA 적응 등화 알고리즘 성능 분석)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.13 no.1
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    • pp.27-34
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    • 2013
  • This paper deals with the performance anlysis of CCA adaptive equalization algorithm, that is used for reduction of intersymbol interference at the receiving side which occurs in the time dispersive communication channel. Basically, this algorithm is borned for the solving phase unrecovery problem in the CMA equalizer, and the comines the concept of DDA (Decision Directed Algorithm) and RCA (Reduce Constellation Algorithm). The DDA has a stable convergence characteristics in unilevel signal, but not in the number of levels in multilevel signal such as QAM, so it has unstable problem. The RCA does not provide reliable initial convergence. And even after convergence, the equalization noise due to the steady state misadjustment exhibited by it is very high as compared to DDA. For the solving the abovemensioned point, the CCA adaptive eualization alogorithm has borned. In order to performance analysis of CCA algorithm, the recovered signal constellation that is the output of the equalizer, the convergence characteristic by the residual isi and MD (maximum distortion), the SER characteristic are used by computer simulation and it was compared with the DDA, RCA respectively. As a result of simulation, the DDA has superior performance than other algoithm, but it has a convergence unguarantee and unstability in the multilevel signal. In order to solving this problem, the CCA has more good performance than RCA in every performance index.

Noise Whitening Decision Feedback Equalizer for SC-FDMA Receivers (SC-FDMA 수신기를 위한 잡음 백색화 판정궤환 등화기)

  • Lee, Su-Kyoung;Park, Yong-Hyun;Seo, Bo-Seok
    • Journal of Broadcast Engineering
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    • v.16 no.6
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    • pp.986-995
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    • 2011
  • In this paper, we propose a noise whitening decision feedback equalizer for single carrier frequency division multiple access (SC-FDMA) receivers. SC-FDMA has the same advantage as that of orthogonal frequency division multiple access (OFDMA) in which the multipath effect can be removed easily, and also solves the problem of high peak to average power ratio (PAPR) which is the main drawback of OFDMA. Although SC-FDMA is a single carrier transmission scheme, a simple frequency domain linear equalizer (FD-LE) can be implemented as in OFDMA, which can dramatically reduce the equalizer complexity. Moreover, some residual intersymbol interference in the output of the FD-LE can be further removed by an additional nonlinear decision feedback equalizer (DFE) in time domain, because the time domain signal is a digitally modulated symbol. In the conventional DFE, however, the noise is not white at the input of the decision device and correspondingly the decision is not optimum. In this paper, we propose an improved DFE scheme for SC-FDMA systems where a linear noise whitening filter is inserted before the decision device of the conventional DFE scheme. Through computer simulations, we compare the bit error rate performance of the proposed DFE scheme with the conventional equalizers.

A Performance Analysis of DF-DPD and DPD-RGPR (DF-DPD와 DPD-RGPR에 대한 성능 분석)

  • Jeong, Jin-Doo;Jin, Yong-Sun;Chong, Jong-Wha
    • 전자공학회논문지 IE
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    • v.47 no.4
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    • pp.39-47
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    • 2010
  • This paper proposes a numerical analysis to prove that the performance of the differential phase detections (DPDs) with the decision feedback, such as the decision feedback DPD (DF-DPD) and the DPD with recursively generated phase reference (DPD-RGPR), approach the performance of the coherent detection with differential decoding. The conventional differential phase detection for M-ary DPSK can make the receiver architecture simple, while it can make the bit-error rate (BER) performance poor because of the previous noisy phase as a reference phase. To improve the BER performance of the conventional differential detection, multiple symbol differential detection methods, including DF-DPD and DPD-RGPR, have been proposed. However, the studies on the analysis and on the comparison of these methods have been little performed. Then, this paper mathematically intends to analyze and compare the performance of the DPDs with the decision feedback. The analysis results show that the DPDs with the decision feedback can have the performance equal to that of the coherent detection with differential decoding and be available for the noncoherent detection in the improved performance. Considering the hardware complexity, the DPD RGPR with the simple detection process by using the recursively generated phase reference can be more simply implemented than the DF-DPD based on the architecture whose complexity increases according to the increasing detection length.

Design of a New 3-D 16-ary Signal Constellation with Constant Envelope (상진폭 특성을 가지는 새로운 3차원 16진 신호성상도의 설계)

  • Choe, Chae-Cheol;Kang, Seog-Geun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.15 no.10
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    • pp.2149-2156
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    • 2011
  • In this paper, design of a new 3-dimensional (3-D) 16-ary signal constellation with constant envelope is presented and analyzed. Unlike the conventional 16-ary constellations, all signal points of the new constellation are uniformly located on the surface of a sphere so that they have a unique amplitude level and a symmetrical structure. When average power of the constellations is normalized, the presented 16-ary constellation has around 11.4% increased minimum Euclidean distance (MED) as compared to the conventional ones that have non-constant envelope. As a result, a digital communication system which exploits the presented constellation has 1.2dB improved symbol error rate (SER). While signal points of the conventional constant-envelope constellation are not distributed uniformly on the surface of a sphere, those of the proposed constellation has a completely symmetric distribution. In addition, the new signal constellation has much lower computational complexity for practical implementation than the conventional one. Hence, the proposed 3-D 16-ary signal constellation is appropriate for the application to a communication system which strongly requires a constant-envelope characteristic.

A 2×2 MIMO Spatial Multiplexing 5G Signal Reception in a 500 km/h High-Speed Vehicle using an Augmented Channel Matrix Generated by a Delay and Doppler Profiler

  • Suguru Kuniyoshi;Rie Saotome;Shiho Oshiro;Tomohisa Wada
    • International Journal of Computer Science & Network Security
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    • v.23 no.10
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    • pp.1-10
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    • 2023
  • This paper proposes a method to extend Inter-Carrier Interference (ICI) canceling Orthogonal Frequency Division Multiplexing (OFDM) receivers for 5G mobile systems to spatial multiplexing 2×2 MIMO (Multiple Input Multiple Output) systems to support high-speed ground transportation services by linear motor cars traveling at 500 km/h. In Japan, linear-motor high-speed ground transportation service is scheduled to begin in 2027. To expand the coverage area of base stations, 5G mobile systems in high-speed moving trains will have multiple base station antennas transmitting the same downlink (DL) signal, forming an expanded cell size along the train rails. 5G terminals in a fast-moving train can cause the forward and backward antenna signals to be Doppler-shifted in opposite directions, so the receiver in the train may have trouble estimating the exact channel transfer function (CTF) for demodulation. A receiver in such high-speed train sees the transmission channel which is composed of multiple Doppler-shifted propagation paths. Then, a loss of sub-carrier orthogonality due to Doppler-spread channels causes ICI. The ICI Canceller is realized by the following three steps. First, using the Demodulation Reference Symbol (DMRS) pilot signals, it analyzes three parameters such as attenuation, relative delay, and Doppler-shift of each multi-path component. Secondly, based on the sets of three parameters, Channel Transfer Function (CTF) of sender sub-carrier number n to receiver sub-carrier number l is generated. In case of n≠l, the CTF corresponds to ICI factor. Thirdly, since ICI factor is obtained, by applying ICI reverse operation by Multi-Tap Equalizer, ICI canceling can be realized. ICI canceling performance has been simulated assuming severe channel condition such as 500 km/h, 8 path reverse Doppler Shift for QPSK, 16QAM, 64QAM and 256QAM modulations. In particular, 2×2MIMO QPSK and 16QAM modulation schemes, BER (Bit Error Rate) improvement was observed when the number of taps in the multi-tap equalizer was set to 31 or more taps, at a moving speed of 500 km/h and in an 8-pass reverse doppler shift environment.

A digital Audio Watermarking Algorithm using 2D Barcode (2차원 바코드를 이용한 오디오 워터마킹 알고리즘)

  • Bae, Kyoung-Yul
    • Journal of Intelligence and Information Systems
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    • v.17 no.2
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    • pp.97-107
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    • 2011
  • Nowadays there are a lot of issues about copyright infringement in the Internet world because the digital content on the network can be copied and delivered easily. Indeed the copied version has same quality with the original one. So, copyright owners and content provider want a powerful solution to protect their content. The popular one of the solutions was DRM (digital rights management) that is based on encryption technology and rights control. However, DRM-free service was launched after Steve Jobs who is CEO of Apple proposed a new music service paradigm without DRM, and the DRM is disappeared at the online music market. Even though the online music service decided to not equip the DRM solution, copyright owners and content providers are still searching a solution to protect their content. A solution to replace the DRM technology is digital audio watermarking technology which can embed copyright information into the music. In this paper, the author proposed a new audio watermarking algorithm with two approaches. First, the watermark information is generated by two dimensional barcode which has error correction code. So, the information can be recovered by itself if the errors fall into the range of the error tolerance. The other one is to use chirp sequence of CDMA (code division multiple access). These make the algorithm robust to the several malicious attacks. There are many 2D barcodes. Especially, QR code which is one of the matrix barcodes can express the information and the expression is freer than that of the other matrix barcodes. QR code has the square patterns with double at the three corners and these indicate the boundary of the symbol. This feature of the QR code is proper to express the watermark information. That is, because the QR code is 2D barcodes, nonlinear code and matrix code, it can be modulated to the spread spectrum and can be used for the watermarking algorithm. The proposed algorithm assigns the different spread spectrum sequences to the individual users respectively. In the case that the assigned code sequences are orthogonal, we can identify the watermark information of the individual user from an audio content. The algorithm used the Walsh code as an orthogonal code. The watermark information is rearranged to the 1D sequence from 2D barcode and modulated by the Walsh code. The modulated watermark information is embedded into the DCT (discrete cosine transform) domain of the original audio content. For the performance evaluation, I used 3 audio samples, "Amazing Grace", "Oh! Carol" and "Take me home country roads", The attacks for the robustness test were MP3 compression, echo attack, and sub woofer boost. The MP3 compression was performed by a tool of Cool Edit Pro 2.0. The specification of MP3 was CBR(Constant Bit Rate) 128kbps, 44,100Hz, and stereo. The echo attack had the echo with initial volume 70%, decay 75%, and delay 100msec. The sub woofer boost attack was a modification attack of low frequency part in the Fourier coefficients. The test results showed the proposed algorithm is robust to the attacks. In the MP3 attack, the strength of the watermark information is not affected, and then the watermark can be detected from all of the sample audios. In the sub woofer boost attack, the watermark was detected when the strength is 0.3. Also, in the case of echo attack, the watermark can be identified if the strength is greater and equal than 0.5.