• Title/Summary/Keyword: fixed-point implementation

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Implementation of the High-Quality Audio System with the Separately Processed Musical Instrument Channels (악기별 분리처리를 통한 고음질 오디오 시스템 구현)

  • Kim, Tae-Hoon;Lee, Sang-Hak;Kim, Dae-Kyung;Lee, Sang-Chan
    • The Journal of the Acoustical Society of Korea
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    • v.32 no.4
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    • pp.346-353
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    • 2013
  • This paper deals with the implementation of a high-quality audio system for karaoke. For improving the key/tempo changes performance, we separated the audio into many musical instrument channels. By separating musical instrument channels, high-quality key/tempo changes can be achieved and we confirmed this using the cross-correlation distribution and the MOS evaluation. The improved audio system was implemented using the TMS320C6747 DSP with fixed/floating-point operations. The implemented audio system can perform the multi-channel WMA decoding, the MP3 encoding/decoding, the wav playing, the EQ, and the key/tempo changes in real time. The WMA channels used for processing the separated instrument channels. The audio system includs the MP3 encoding/decoding function for playing and recording and the wav channel for the effect sound.

On a Speech Coding Algorithm for Low Cost Implementation of Voice Telegram System (보이스 전보 시스템 구현을 위한 저가형 음성파형 부호화 알고리즘)

  • 나덕수;민소연;배명진
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.2
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    • pp.101-105
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    • 2000
  • A telegram has been used to transmit the emergency news or celebration message. So, it has been very important media in our life. Although the telegram processing is more and more convenient, on the other hand, the telegram service contains only text message. The voice telegram is that delivering user's voice with text message. So, the voice telegram can be delivered sender's emotions and feelings. However, since voice information contains lots of data, large memory size and high cost processor are needed to deliver itself. In this paper, we proposed a new speech waveform coding method that has low complexity and low cost implementation for the voice telegram system. First, we fixed one basic speech waveform per pitch period and measured the waveform similarity between basic and neighbor speech waveform. Second, if the similarity satisfied threshold values, we compress the neighbor speech waveform with pitch and magnitude value per pitch period and if not, we save speech waveform. When the compression is about 45%, we obtained about 4 point in MOS.

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Hardware Design of Elliptic Curve processor Resistant against Simple Power Analysis Attack (단순 전력분석 공격에 대처하는 타원곡선 암호프로세서의 하드웨어 설계)

  • Choi, Byeong-Yoon
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.16 no.1
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    • pp.143-152
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    • 2012
  • In this paper hardware implementation of GF($2^{191}$) elliptic curve cryptographic coprocessor which supports 7 operations such as scalar multiplication(kP), Menezes-Vanstone(MV) elliptic curve cipher/decipher algorithms, point addition(P+Q), point doubling(2P), finite-field multiplication/division is described. To meet structure resistant against simple power analysis, the ECC processor adopts the Montgomery scalar multiplication scheme which main loop operation consists of the key-independent operations. It has operational characteristics that arithmetic units, such GF_ALU, GF_MUL, and GF_DIV, which have 1, (m/8), and (m-1) fixed operation cycles in GF($2^m$), respectively, can be executed in parallel. The processor has about 68,000 gates and its simulated worst case delay time is about 7.8 ns under 0.35um CMOS technology. Because it has about 320 kbps cipher and 640 kbps rate and supports 7 finite-field operations, it can be efficiently applied to the various cryptographic and communication applications.

Implementation of Precise Level Measurement Device using Zoom FFT (Zoom FFT를 이용한 정밀 레벨 측정 장치의 구현)

  • Ji, Suk-Joon;Lee, John-Tark
    • Journal of Advanced Marine Engineering and Technology
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    • v.36 no.4
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    • pp.504-511
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    • 2012
  • In this paper, level instrument is implemented using beat frequency for distance measurement which means the difference between Tx and Rx signal frequency from FMCW Radar Level Transmitter. Beat frequency is analyzed through Fast Fourier Transform of which frequency precision can be improved by applying Zoom FFT. Distance precision is improved from 146.5[mm] to 5[mm] using the advantage of Zoom FFT which can raise the frequency precision without changing the sampling frequency or FFT point number to be fixed in the beginning of designing signal processing. Also, measurement error can be reduced within 2[mm] by incresing the FFT points using the method of Spline interpolation. For verifying the effectiveness of this Zoom FFT to FMCW Radar Level Transmitter, test bench is made to measure the distance for every 1[mm] between 700[mm] and 2000[mm] and measurement error can be checked in the range of ${\pm}2$[mm].

Design and Implementation of an IP-based Fixed VoIP Emergency System (IP-기반 고정형 VoIP 긴급통화 시스템 설계 및 구현)

  • Ko, Sang-Ki;Chon, Ji-Hun;Choi, Sun-Wan;Kang, Shin-Gak;Huh, Mi-Young
    • The KIPS Transactions:PartC
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    • v.15C no.4
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    • pp.245-252
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    • 2008
  • An emergency service over Voice over IP (VoIP) network is an essential condition, like the existing telecommunication services. To support for the emergency services, standardization works have been performed. The National Emergency Number Association (NENA) has been developing the framework and procedures for an emergency service for Non-IP based network, rather than protocols. In contrast, the Internet Engineering Task Force (IETF) has been only focused on end-to-end IP-based emergency calls. The NENA architecture is incompatible with the IETF protocols. To solve the problem, we design and implement a SIP-based VoIP emergency system by adopting the NENA architecture and by applying IETF protocols, for both IP-based Pubic Safety Answering Point (PSAP) and PSTN-based PSAP. It is implemented and tested under UNIX environment.

Vehicle ECU Design Incorporating LIN/CAN Vehicle Interface with Kalman Filter Function (LIN/CAN 차량용 인터페이스와 칼만 필터 기능을 통합한 차량용 ECU 설계)

  • Jeong, Seonwoo;Kim, Yongbin;Lee, Seongsoo
    • Journal of IKEEE
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    • v.25 no.4
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    • pp.762-765
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    • 2021
  • In this paper, an automotive ECU (electronic control unit) with Kalman filter accelerator is designed and implemented. RISC-V is exploited as a processor core. Accelerator for Kalman filter matrix operation, CAN (controller area network) controller for in-vehicle network, and LIN (local interconnect network) controller are designed and embedded. Kalman filter operation consists of time update process and measurement update process. Current state variable and its error covariance are estimated in time update process. Final values are corrected from input measurement data and Kalman gain in measurement update process. Usually floating-point multiplication is exploited in software implementation, but fixed-point multiplier considering accuracy analysis is exploited in this paper to reduce hardware area. In 28nm silicon fabrication, its operating frequency, area, and gate counts are 100MHz, 0.37mm2, and 760k gates, respectively.

Time-optimized Color Conversion based on Multi-mode Chrominance Reconstruction and Operation Rearrangement for JPEG Image Decoding (JPEG 영상 복원을 위한 다중 모드 채도 복원과 연산 재배열 기반의 시간 최적화된 컬러 변환)

  • Kim, Young-Ju
    • Journal of the Korea Society of Computer and Information
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    • v.14 no.1
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    • pp.135-143
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    • 2009
  • Recently, in the mobile device, the increase of the need for encoding and decoding of high-resolution images requires an efficient implementation of the image codec. This paper proposes a time-optimized color conversion method for the JPEG decoder, which reduces the number of calculations in the color conversion by the rearrangement of arithmetic operations being possible due to the linearity of the IDCT and the color conversion matrices and brings down the time complexity of the color conversion itself by the integer mapping replacing floating-point operations to the optimal fixed-point shift and addition operations, eventually reducing the time complexity of the JPEG decoder. And the proposed method compensates a decline of image quality incurred by the quantification error of the operation arrangement and the integer mapping by using the multi-mode chrominance reconstruction. The performance evaluation performed on the development platform of embedded systems showed that, compared to previous color conversion methods, the proposed method greatly reduces the image decoding time, minimizing the distortion of decoded images.

Implementation of Exchange Rate Forecasting Neural Network Using Heterogeneous Computing (이기종 컴퓨팅을 활용한 환율 예측 뉴럴 네트워크 구현)

  • Han, Seong Hyeon;Lee, Kwang Yeob
    • Asia-pacific Journal of Multimedia Services Convergent with Art, Humanities, and Sociology
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    • v.7 no.11
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    • pp.71-79
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    • 2017
  • In this paper, we implemented the exchange rate forecasting neural network using heterogeneous computing. Exchange rate forecasting requires a large amount of data. We used a neural network that could leverage this data accordingly. Neural networks are largely divided into two processes: learning and verification. Learning took advantage of the CPU. For verification, RTL written in Verilog HDL was run on FPGA. The structure of the neural network has four input neurons, four hidden neurons, and one output neuron. The input neurons used the US $ 1, Japanese 100 Yen, EU 1 Euro, and UK £ 1. The input neurons predicted a Canadian dollar value of $ 1. The order of predicting the exchange rate is input, normalization, fixed-point conversion, neural network forward, floating-point conversion, denormalization, and outputting. As a result of forecasting the exchange rate in November 2016, there was an error amount between 0.9 won and 9.13 won. If we increase the number of neurons by adding data other than the exchange rate, it is expected that more precise exchange rate prediction will be possible.

Optimized DSP Implementation of Audio Decoders for Digital Multimedia Broadcasting (디지털 방송용 오디오 디코더의 DSP 최적화 구현)

  • Park, Nam-In;Cho, Choong-Sang;Kim, Hong-Kook
    • Journal of Broadcast Engineering
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    • v.13 no.4
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    • pp.452-462
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    • 2008
  • In this paper, we address issues associated with the real-time implementation of the MPEG-1/2 Layer-II (or MUSICAM) and MPEG-4 ER-BSAC decoders for Digital Multimedia Broadcasting (DMB) on TMS320C64x+ that is a fixed-point DSP processor with a clock speed of 330 MHz. To achieve the real-time requirement, they should be optimized in different steps as follows. First of all, a C-code level optimization is performed by sharing the memory, adjusting data types, and unrolling loops. Next, an algorithm level optimization is carried out such as the reconfiguration of bitstream reading, the modification of synthesis filtering, and the rearrangement of the window coefficients for synthesis filtering. In addition, the C-code of a synthesis filtering module of the MPEG-1/2 Layer-II decoder is rewritten by using the linear assembly programming technique. This is because the synthesis filtering module requires the most processing time among all processing modules of the decoder. In order to show how the real-time implementation works, we obtain the percentage of the processing time for decoding and calculate a RMS value between the decoded audio signals by the reference MPEG decoder and its DSP version implemented in this paper. As a result, it is shown that the percentages of the processing time for the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders occupy less than 3% and 11% of the DSP clock cycles, respectively, and the RMS values of the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders implemented in this paper all satisfy the criterion of -77.01 dB which is defined by the MPEG standards.

Implementation of Internet Terminal using G.729.1 Wideband Speech Codec for Next Generation Network (차세대 통신망을 위한 G.729.1 광대역 음성 코덱을 활용한 인터넷 단말 구현)

  • So, Woon-Seob;Kim, Dae-Young
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.10B
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    • pp.939-945
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    • 2008
  • Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.