• Title/Summary/Keyword: DSP implementation

검색결과 701건 처리시간 0.022초

멀티채널 AMR 음성부호화기의 실시간 구현 (Real-time Implementation of Multi-channel AMR Speech Coder)

  • 지덕구;박만호;김형중;윤병식;최송인
    • 한국음향학회지
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    • 제20권8호
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    • pp.19-23
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    • 2001
  • 고속 저전력의 DSP (Programmable Digital Signal Processor)가 개발됨에 따라 이동통신 분야에서 시스템 및 단말기 등이 DSP를 사용하여 구현되고 있다. 본 논문에서는 DSP를 사용한 AMR (Adaptive Multi-rate) 음성부호화기의 멀티 채널 실시간 구현에 관하여 논한다. AMR 음성부호화 알고리즘을 250 MHz로 동작하는 32비트 정수형 DSP 칩인 TMS320C6202를 사용하여 구현하였다. 실시간 동작을 위하여 cross compile, 선형 어셈블리 최적화, TMS320C62xx 어셈블리 최적화 작업을 수행하였다. AMR 음성부호화기에 음성 데이터 입출력 기능 및 외부 CPU와의 통신기능을 포함하였다. DSP EVM 보드를 사용하여 AMR 음성부호화기를 개발하였고, ETRI에서 개발중인 비동기 IMT-2000 시스템 상에서 동작 및 기능을 검증하였다.

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Low Power DSP Implementation of 3D Sound Localization

  • Sakamoto, Noriaki;Kobayashi, Wataru;Onoye, Takao;Shirakawa, Isao
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2000년도 ITC-CSCC -1
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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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TMS320C6670 기반 LTE-A PDSCH 디코더 구현 (Implementation of LTE-A PDSCH Decoder using TMS320C6670)

  • 이광민;안흥섭;최승원
    • 디지털산업정보학회논문지
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    • 제14권4호
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    • pp.79-85
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    • 2018
  • This paper presents an implementation method of Long Term Evolution-Advanced (LTE-A) Physical Downlink Shared Channel (PDSCH) decoder using a general-purpose multicore Digital Signal Processor (DSP), TMS320C6670. Although the DSP provides some useful coprocessors such as turbo decoder, fast Fourier transformer, Viterbi Coprocessor, Bit Rate Coprocessor etc., it is specific to the base station platform implementation not the mobile terminal platform implementation. This paper shows an implementation method of the LTE-A PDSCH decoder using programmable DSP cores as well as the coprocessors of Fast Fourier Transformer and turbo decoder. First, it uses the coprocessor supported by the TMS320C6670, which can be used for PDSCH implementation. Second, we propose a core programming method using DSP optimization method for block diagram of PDSCH that can not use coprocessor. Through the implementation, we have verified a real-time decoding feasibility for the LTE-A downlink physical channel using test vectors which have been generated from LTE-A Reference Measurement Channel (RMC) Waveform R.6.

FPGA와 DSP를 이용한 실시간 차선 및 차량인식 시스템 구현 (FPGA-DSP Based Implementation of Lane and Vehicle Detection)

  • 김일호;김경환
    • 한국통신학회논문지
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    • 제36권12C호
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    • pp.727-737
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    • 2011
  • 본 논문에서는 FPGA(Field Programmable Gate Array)와 DSP(Digital Signal Processor)를 이용하는 실시간 차선 및 차량인식 시스템의 구현에 대하여 기술한다. 실시간 시스템의 구현을 위해서 FPGA와 DSP의 역할을 효율적으로 분할할 필요성이 있다. 시스템의 알고리즘을 특정요소 추출부분을 기준으로 분할하여 대량의 영상정보를 이용하여 소량의 특정요소를 추출하는 과정을 FPGA로 구현하고 추출된 특정요소를 사용하여 차선과 차량을 정의하고 추적하는 부분을 DSP에서 수행하게 하고, FPGA와 DSP의 효율적 연동을 위한 인터페이스 구성을 제안함으로써 실시간 처리가 가능한 시스템 구조를 제안한다. 실험 결과 제안한 실시간 차선 및 차량인식 시스템은 $640{\times}480$ 크기를 갖는 비디오 영상 입력에 대해 약 15 (frames/sec)로 동작하여 실시간 응용으로 충분함을 알 수 있다.

DSP56362를 이용한 G.723.1 음성코덱의 실시간 구현 (Real-time implementation of the G.723.1 voice coder using DSP56362)

  • 이재식;손용기;장태규;민병기
    • 음성과학
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    • 제7권2호
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    • pp.225-234
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    • 2000
  • This paper describes the fixed-point DSP implementation of a CELP(Code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56362. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.

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멀티코어 DSP를 사용한 SDR 기반 IEEE 802.11ac 인코더의 설계 및 구현 (Design and Implementation of Software Defined Radio Based IEEE 802.11ac Encoder Using Multicore DSP)

  • 장중봉;안흥섭;최승원
    • 디지털산업정보학회논문지
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    • 제15권4호
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    • pp.93-101
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    • 2019
  • This paper presents a software design and implementation of software-defined radio based IEEE 802.11ac encoder using Texas Instruments TMS320C6670 digital signal processor (DSP) platform. In this paper, the implemented encoder has the capability of generating all the signals consisting of preamble field and data field under different modulation & coding scheme in the IEEE 802.11ac standard. Moreover, the flexibility in choosing different rate, bandwidth, or mode can also be achieved by software reconfiguration using the DSP. As a result, by utilizing the computing power provided by multi-cores as well as the FFT coprocessors in the DSP, the required maximum throughput 78Mbps can be fully reached within 4 ㎲ for each OFDM symbol in the case of 20MHz bandwidth of IEEE 802.11ac.

TMS320F28335 DSP를 이용한 화자독립 음성인식기 구현 (Implementation of a Speaker-independent Speech Recognizer Using the TMS320F28335 DSP)

  • 정익주
    • 산업기술연구
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    • 제29권A호
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    • pp.95-100
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    • 2009
  • In this paper, we implemented a speaker-independent speech recognizer using the TMS320F28335 DSP which is optimized for control applications. For this implementation, we used a small-sized commercial DSP module and developed a peripheral board including a codec, signal conditioning circuits and I/O interfaces. The speech signal digitized by the TLV320AIC23 codec is analyzed based on MFCC feature extraction methed and recognized using the continuous-density HMM. Thanks to the internal SRAM and flash memory on the TMS320F28335 DSP, we did not need any external memory devices. The internal flash memory contains ADPCM data for voice response as well as HMM data. Since the TMS320F28335 DSP is optimized for control applications, the recognizer may play a good role in the voice-activated control areas in aspect that it can integrate speech recognition capability and inherent control functions into the single DSP.

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멀티코어 DSP를 이용한 다중 안테나를 지원하는 SDR 기반 LTE-A PDSCH 디코더 구현 (Implementation of SDR-based LTE-A PDSCH Decoder for Supporting Multi-Antenna Using Multi-Core DSP)

  • 나용;안흥섭;최승원
    • 디지털산업정보학회논문지
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    • 제15권4호
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    • pp.85-92
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    • 2019
  • This paper presents a SDR-based Long Term Evolution Advanced (LTE-A) Physical Downlink Shared Channel (PDSCH) decoder using a multicore Digital Signal Processor (DSP). For decoder implementation, multicore DSP TMS320C6670 is used, which provides various hardware accelerators such as turbo decoder, fast Fourier transformer and Bit Rate Coprocessors. The TMS320C6670 is a DSP specialized in implementing base station platforms and is not an optimized platform for implementing mobile terminal platform. Accordingly, in this paper, the hardware accelerator was changed to the terminal implementation to implement the LTE-A PDSCH decoder supporting the multi-antenna and the functions not provided by the hardware accelerator were implemented through core programming. Also pipeline using multicore was implemented to meet the transmission time interval. To confirm the feasibility of the proposed implementation, we verified the real-time decoding capability of the PDSCH decoder implemented using the LTE-A Reference Measurement Channel (RMC) waveform about transmission mode 2 and 3.

ACELP/MP-MLQ에 기초한 dual-rate 음성 코더의 DSP 구현 (Implementation of the ACELP/MPMLQ-Based Dual-Rate Voice Coder Using DSP)

  • 이재식;손용기;전일;장태규;민병기
    • 한국음향학회:학술대회논문집
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    • 한국음향학회 2000년도 하계학술발표대회 논문집 제19권 1호
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    • pp.51-54
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    • 2000
  • This paper describes the fixed-point DSP implementation of a CELP(code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically Parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56309. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.

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DSP 프로세서를 이용한 실시간 ANC 시스템 구현에 관한 연구 (Implementation of Real-Time Adaptive Noise Cancellation System Using DSP Processor)

  • 이영일;최홍섭
    • 대한음성학회지:말소리
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    • 제52호
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    • pp.121-132
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    • 2004
  • This paper is aiming at real-time implementation of adaptive noise cancellation system using DSP processor. ACHARF algorithm, which guarantees stability and fast convergence by adaptive compensator, is used on this DSP system. For the experiments, TLV320AIC23 stereo CODEC of TI Inc. is used with TMS320C6413 DSP processor. Signals of primary input and reference input are obtained by two microphones. The primary input is the voice plus noise signal and the reference input is white noise or real noise. The experimental results show that ANC system using DSP processor with ACHARF is verified to be an effective speech enhancement method for various speech processing units.

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