• Title/Summary/Keyword: 16-bit fixed-point

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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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Real-Time Implementation of AMR Speech Codec Using TMS320VC5510 DSP (TMS320VC5510 DSP를 이용한 AMR 음성부호화기의 실시간 구현)

  • Kim, Jun;Bae, Keun-Sung
    • MALSORI
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    • no.65
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    • pp.143-152
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    • 2008
  • This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

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A Fixed-point implementation of MPEG-4 CELP coder (고정소수점 연산구조에 기초한 MPEG-4 CELP coder구현)

  • 이우종;이재식;박지태;장태규;이전우
    • Proceedings of the IEEK Conference
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    • 2001.06d
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    • pp.119-122
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    • 2001
  • 본 논문에서는 음성압축 앨고리즘인 MPEG-4 CELP coder를 16 bit DSP 구현에 필요한 고정소수점 연산구조로 구현하였다. 기본 앨고리즘 중에 LSP 계수를 구하는 방법인 Chebyshev series method 대신 고정소수점 구현에 유리한 Real root method 앨고리즘을 사용하였다. 또한 cosine, log 둥 DSP 명령어가 지원하지 않는 수학 함수들은 미리 계산하여 테이블 적용기법을 사용하였고 고정 소수점 연산에 불리한 나눗셈 연산을 최대한 배제하였다. 고정 소수점 연산 구조로 변환한 후 부동 소수점 연산구조와의 비교를 통하여 오차를 최소화하도록 하였다 구현한 음성코더를 남, 여 각 5문장에 적용했을 때 부동 소수점 연산구조에 비교해 음질의 열화가 없음을 확인하였다.

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Low Power DSP Implementation of 3D Sound Localization

  • Sakamoto, Noriaki;Kobayashi, Wataru;Onoye, Takao;Shirakawa, Isao
    • Proceedings of the IEEK Conference
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    • 2000.07a
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    • pp.253-256
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    • 2000
  • This paper describes a DSP implementation of a real-time 3D sound localization algorithm with the use of a low power embedded DSP. A distinctive feature of this implementation is that the audible frequency band is divided into three, in accordance with the sound reflection and diffraction phenomena through different media from a certain sound source to human ears, and then in each subband a specific implementation procedure of the 3D sound localization is devised so as to operate real-time at a low frequency of 50MHz on a 16bit fixed-point DSP. Thus out DSP implementation can provide a listener with 3D sound effects through a headphone at low cost and low power consumption.

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Implementation of a Real-time SIFT Pitch Detector (실시간 SIFT 기본주파수 검출기의 구현)

  • Lee, Jong Seok;Lee, Sang Uk
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.23 no.1
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    • pp.101-113
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    • 1986
  • In this paper, a real-time pitch detector LPC vocoder as implemented on a high speed digital signal processor, NEC 7720, is described. The pitch detector was based mainly on the SIFT algorithm. The SIFT pitch detector consists primarily of a digital low pass filter, inverse filter, computation of autocorrelation, a peak picker, interpolation, V/UV defcision and a final pitch smoother. In our approach, modification, mainly on the V/UV decision and a final pitch smoother, was made to estimate more accurate pitches. An 16-bit fixed-point aithmatic was employed for all necessary computation and the simulated results were compared with the eye detected pitches obtained from real speech data. The pitch detector occupies 98.8% of the instruction ROM, 37% of the data ROM, and 94% of internal RAM and takes 15.2ms to estimate a pitch when an analysis frame is consisted of 128 sampled speech data. It is observed that the tested results were well agreed with the computer simulation results.

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A Design of High-speed Phase Calculator for 3D Depth Image Extraction from TOF Sensor Data (TOF 센서용 3차원 Depth Image 추출을 위한 고속 위상 연산기 설계)

  • Koo, Jung-Youn;Shin, Kyung-Wook
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.17 no.2
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    • pp.355-362
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    • 2013
  • A hardware implementation of phase calculator for extracting 3D depth image from TOF(Time-Of-Flight) sensor is described. The designed phase calculator, which adopts a pipelined architecture to improve throughput, performs arctangent operation using vectoring mode of CORDIC algorithm. Fixed-point MATLAB modeling and simulations are carried out to determine the optimized bit-widths and number of iteration. The designed phase calculator is verified by FPGA-in-the-loop verification using MATLAB/Simulink, and synthesized with a TSMC 0.18-${\mu}m$ CMOS cell library. It has 16,000 gates and the estimated throughput is about 9.6 Gbps at 200Mhz@1.8V.

A Single-Chip Video/Audio CODEC for Low Bit Rate Application

  • Park, Seong-Mo;Kim, Seong-Min;Kim, Ig-Kyun;Byun, Kyung-Jin;Cha, Jin-Jong;Cho, Han-Jin
    • ETRI Journal
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    • v.22 no.1
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    • pp.20-29
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    • 2000
  • In this paper, we present a design of video and audio single chip encoder/decoder for portable multimedia application. The single-chip called as video audio signal processor (VASP) consists of a video signal processing block and an audio single processing block. This chip has mixed hardware/software architecture to combine performance and flexibility. We designed the chip by partitioning between video and audio block. The video signal processing block was designed to implement hardware solution of pixel input/output, full pixel motion estimation, half pixel motion estimation, discrete cosine transform, quantization, run length coding, host interface, and 16 bits RISC type internal controller. The audio signal processing block is implemented with software solution using a 16 bits fixed point DSP. This chip contains 142,300 gates, 22 Kbits FIFO, 107 kbits SRAM, and 556 kbits ROM, and the chip size is $9.02mm{\times}9.06mm$ which is fabricated using 0.5 micron 3-layer metal CMOS technology.

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Implementation of Adaptive Multi Rate (AMR) Vocoder for the Asynchronous IMT-2000 Mobile ASIC (IMT-2000 비동기식 단말기용 ASIC을 위한 적응형 다중 비트율 (AMR) 보코더의 구현)

  • 변경진;최민석;한민수;김경수
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.1
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    • pp.56-61
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    • 2001
  • This paper presents the real-time implementation of an AMR (Adaptive Multi Rate) vocoder which is included in the asynchronous International Mobile Telecommunication (IMT)-2000 mobile ASIC. The implemented AMR vocoder is a multi-rate coder with 8 modes operating at bit rates from 12.2kbps down to 4.75kbps. Not only the encoder and the decoder as basic functions of the vocoder are implemented, but VAD (Voice Activity Detection), SCR (Source Controlled Rate) operation and frame structuring blocks for the system interface are also implemented in this vocoder. The DSP for AMR vocoder implementation is a 16bit fixed-point DSP which is based on the TeakLite core and consists of memory block, serial interface block, register files for the parallel interface with CPU, and interrupt control logic. Through the implementation, we reduce the maximum operating complexity to 24MIPS by efficiently managing the memory structure. The AMR vocoder is verified throughout all the test vectors provided by 3GPP, and stable operation in the real-time testing board is also proved.

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Design of a High-speed Decision Feedback Equalizer using the Constant-Modulus Algorithm (CMA 알고리즘을 이용한 고속 DFE 등화기 설계)

  • Jeon, Yeong-Seop;;Kim, Gyeong-Ho
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.39 no.4
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    • pp.173-179
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    • 2002
  • This paper describes an equalizer using the DFE (Decision Feedback Equalizer) structure, CMA (Constant Modulus Algorithm) and LMS (Least Mean Square) algorithms. The DFE structure has better channel adaptive performance and lower BER than the transversal structure. The proposed equalizer can be used for 16/64 QAM modems. We employ high speed multipliers, square logics and many CSAs (Carry Save Adder) for high speed operations. We have developed floating-point models and fixed-point models using the COSSAP$\^$TM/ CAD tool and developed VHDL filter. The proposed equalizer shows low BER in multipath fading channel. We have performed models. From the simulation results, we employ a 12 tap feedback filter and a 8 tap feedforward logic synthesis using the SYNOPSYS$\^$TM/ CAD tool and the SAMSUNG 0.5$\mu\textrm{m}$ standard cell library (STD80) and verified function and timing simulations. The total number of gates is about 130,000.

Repetitive Control for Track-Following Servo of an Optical Disk Drive Using Linear Matrix Inequalities (선형 행렬 부등식을 이용한 광 디스크 드라이브의 트랙 추종 서보를 위한 반복 제어)

  • 도태용;문정호
    • Journal of Institute of Control, Robotics and Systems
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    • v.9 no.2
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    • pp.117-123
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    • 2003
  • Rotational machines such as optical disk drives, hard disk drives, and so on are subject to periodic disturbances caused by their mechanical characteristics. In the meanwhile, it is well known that repetitive control rejects periodic disturbance effectively. This paper presents a practical application of repetitive control to the track-following servo of an optical disk drive. The repetitive control system is composed of two repetitive controllers which compensate for periodic disturbances generated by track geometry and eccentric rotation of disk and a feedback controller stabilizing the feedback loop. A robust stability for all plant uncertainties is proved using linear matrix inequalities (LMIs). In the controller design, a weighting function is introduced for the feedback controller to ensure a minimum loop gain and a sufficient phase margin. The repetitive controllers and the feedback controller are designed by solving an optimization problem which can consider the robust stability condition and the system performance. The developed repetitive control system is implemented in the digital control system with a 16-bit fixed-point digital signal processor (DSP). Through simulation and experiment. The feasibility of the proposed repetitive control system is verified.