• Title/Summary/Keyword: fixed-point implementation

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Multi-mode Layered LDPC Decoder for IEEE 802.11n (IEEE 802.11n용 다중모드 layered LDPC 복호기)

  • Na, Young-Heon;Shin, Kyung-Wook
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.48 no.11
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    • pp.18-26
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    • 2011
  • This paper describes a multi-mode LDPC decoder which supports three block lengths(648, 1296, 1944) and four code rates(1/2, 2/3, 3/4, 5/6) of IEEE 802.11n wireless LAN standard. To minimize hardware complexity, it adopts a block-serial (partially parallel) architecture based on the layered decoding scheme. A novel memory reduction technique devised using the min-sum decoding algorithm reduces the size of check-node memory by 47% as compared to conventional method. From fixed-point modeling and Matlab simulations for various bit-widths, decoding performance and optimal hardware parameters such as fixed-point bit-width are analyzed. The designed LDPC decoder is verified by FPGA implementation, and synthesized with a 0.18-${\mu}m$ CMOS cell library. It has 219,100 gates and 45,036 bits RAM, and the estimated throughput is about 164~212 Mbps at 50 MHz@2.5v.

Implementation of Acoustic Echo Canceller with A Post-processor Using A Fixed-Point DSP (고정 소수점 DSP를 이용한 후처리기를 가지는 음향 반향제거기의 구현)

  • 이영호;박장식;박주성;손경식
    • Journal of Korea Multimedia Society
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    • v.3 no.3
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    • pp.263-271
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    • 2000
  • In this paper, an acoustic echo canceller(AEC) is implemented by ADSP-2181. This AEC uses a noise robust adaptive algorithm and a postprocessing method which attenuates residual echo using cross-correlation between estimated error signal and microphone input signal. We propose new postprocessing method that uses two thresholds to prevent signal distortion after postprocessing and to improve the performance of AEC without extra computational burden. Through experiments using a 16 bit fixed-point DSP board (ADSP-2181 EZ-KIT Lite board), it is shown that the noise robust adaptive algorithm performs well in the double-talk situations and the convergence speed is comparable to NLMS. Using the postprocessor, ERLE is improved about 20 dB. As a result, the AEC with a postprocessor shows better performance than conventional ones.

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A performance analysis of LDPC decoder for IEEE 802.16e WiMAX System (IEEE 802.16e WiMAX용 LDPC 복호기의 성능분석)

  • Kim, Eun-Suk;Kim, Hae-Ju;Shin, Kyung-Wook
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2010.10a
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    • pp.722-725
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    • 2010
  • In this paper, BER performance and error convergence speed of layered LDPC(Low Density Parity Check) decoder which supports IEEE 802.16e WiMAX standard is analyzed, and optimal design conditions for hardware implementation are derived. A LDPC decoder is modeled and simulated at AWGN channel with QPSK modulation by Matlab. The parity check matrix(PCM) for IEEE 802.16e standard which has block lengths of 576, 1440, 2304 and code rates of 1/2, 2/3A, 2/3B, 3/4A, 3/4B, 5/6 are used. Fixed-point simulation results show that fixed-point bit-width should be more than 8 bits for acceptable decoding performance.

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Performance Analysis of MlMO-OFDMA System Combined with Adaptive Beamforming (다중 입출력과 적응형 빔형성 기술 결합기법을 적용한 직교주파수분할 다중 접속시스템의 성능 분석)

  • Chung, Jae-Ho;Choi, Seung-Won
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.2C
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    • pp.86-92
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    • 2011
  • This paper details the downlink performance analysis of an multiple antennas system that combines adaptive beamforming and spatial multiplexing (SM) Multiple Input Multiple Output (MIMO). The combination of MIMO signal processing with adaptive beamforming is applied to WiBro, the South Korean Orthogonal Frequency Division Multiple Access (OFDMA) system that follows the IEEE 802.16e standard. Performance analysis is based on the results of experiments and simulations obtained from a fixed-point simulation testbed. Simulations demonstrate that the MIMO Beamforming OFDMA system improves the required signal to noise ratio (SNR) over the conventional MIMO OFDMA system by 3 dB (QPSK) / 2.5 dB (16-QAM) for the frame error rate (FER) of 1% in the WiBro signal environments. From the implementation of the fixed-point simulation testbed and its experimental results, we verify the feasibility of the MIMO Beamforming technology for realizing a practical WiBro base station.

Performance Analysis of Error Correction Codes for 3GPP Standard (3GPP 규격 오류 정정 부호 기법의 성능 평가)

  • 신나나;이창우
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.15 no.1
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    • pp.81-88
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    • 2004
  • Turbo code has been adopted in the 3GPP standard, since its performance is very close to the Shannon limit. However, the turbo decoder requires a lot of computations and the amount of the memory increases as the block size of turbo codes becomes larger. In order to reduce the complexity of the turbo decoder, the Log-MAP, the Max-Log-MAP and the sliding window algorithm have been proposed. In this paper, the performance of turbo codes adopted in the 3GPP standard is analyzed by using the floating point and the fixed point implementation. The efficient decoding method is also proposed. It is shown that the BER performance of the proposed method is close to that of the Log-MAP algorithm.

Synthesis of 3D Sound Movement by Embedded DSP

  • Komata, Shinya;Sakamoto, Noriaki;Kobayashi, Wataru;Onoye, Takao;Shirakawa, Isao
    • Proceedings of the IEEK Conference
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    • 2002.07a
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    • pp.117-120
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    • 2002
  • A single DSP implementation of 3D sound movement is described. With the use of a realtime 3D acoustic image localization algorithm, an efficient approach is devised for synthesizing the 3D sound movement by interpolating only two parameters of "delay" and "gain". Based on this algorithm, the realtime 3D sound synthesis is performed by a commercially available 16-bit fixed-point DSP with computational labor of 65 MIPS and memory space of 9.6k words, which demonstrates that the algorithm call be used even for the mobile applications.

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Adaptive Multi-Rate(AMR) Speech Coding Algorithm (Adaptive Multi-Rate(AMR) 음성부호화 알고리즘)

  • 서정욱;배건성
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.92-97
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    • 2000
  • An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

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Real-time Implementation of 2.4kbps MELP vocoder using the TMS320C542 (TMS320C542를 이용한 2.4kbps MELP 보코더의 실시간 구현)

  • Park Young-Ho;Jung Chan-Joong;Bae Myung-Jin
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.145-148
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    • 2000
  • 본 논문은 범용 16bit Fixed-point DSP를 이용한 새로운 미국 DoD 2.4kbps MELP(Mixed Excitation Linear Predictive)보코더의 실시간 구현에 관한 것이다. 구현된 MELP보코더는 ROM 32.6kword, RAM 12.2kword를 가지며 40MIPS DSP에서 약 29MIPS를 필요로 하였다. 출력된 파형은 C simulator 와 Bit Exact한 출력 결과를 보여주었다. 실시간 구현된 MELP를 동일전송율의 2.4kbps AMBE와 음질 비교한 결과 AME보다는 MOS 0.2 음질 이 떨어졌다

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Implementation of the Multi-Segment Karatsuba Multiplier for Binary Field (멀티 세그먼트 카라츄바 유한체 곱셈기의 구현)

  • Oh, Jong-Soo
    • Proceedings of the KIEE Conference
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    • 2004.11c
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    • pp.129-131
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    • 2004
  • Elliptic Curve Cryptography (ECC) coprocessors support massive scalar multiplications of a point. We research the design for multi-segment multipliers in fixed-size ECC coprocessors using the multi-segment Karatsuba algorithm on GF($2^m$). ECC coprocessors of the proposed multiplier is verified on the SoC-design verification kit which embeds ALTERA EXCALIBUR FPGAs. As a result of our experiment, the multi-segment Karatsuba multiplier, which has more efficient performance about twice times than the traditional multi-segment multiplier, can be implemented as adding few H/W resources. Therefore the multi-segment Karatsuba multiplier which satisfies performance for the cryptographic algorithm, is adequate for a low cost embedded system, and is implemented in the minimum area.

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Architecture of 2-D DCT processor adopting accuracy comensator (정확도 보상기를 적용한 2차원 이산 코사인 변환 프로세서의 구조)

  • 김견수;장순화;김재호;손경식
    • Journal of the Korean Institute of Telematics and Electronics A
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    • v.33A no.10
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    • pp.168-176
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    • 1996
  • This paper presetns a 2-D DCT architecture adopting accurac y compensator for reducing the hardware complexity and increasing processing speed in VL\ulcornerSI implementation. In the application fields such as moving pictures experts group (MPEG) and joint photographic experts group (JPEG), 2-D DCT processor must be implemented precisely enough to meet the accuracy specifications of the ITU-T H.261. Almost all of 2-D DCT processors have been implemented using many multiplications and accumulations of matrices and vectors. The number of multiplications and accumulations seriously influence on comlexity and speed of 20D DCT processor. In 2-D DCT with fixed-point calculations, the computation bit width must be sufficiently large for the above accuracy specifications. It makes the reduction of hardware complexity hard. This paper proposes the accuracy compensator which compensates the accuracy of the finite word length calculation. 2-D DCT processor with the proposed accuracy compensator shows fairly reduced hardware complexity and improved processing speed.

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