• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.2
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• pp.41-52
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• 2011

It is important to reduce computational complexity of de-blocking filter for real-time implementation, because it accounts for a great part of total computational complexity of the decoder. Because there are a lot of conditional branches and memory accesses in a decoding loop, it is not easy to speed up the de-blocking filter. Therefore, this paper presents a new algorithm of de-blocking filter minimizing conditional branches and memory accesses. The proposed structure of de-blocking filter enables filter operation to parallelize by software pipelining. The proposed optimization method was implemented on a TMS320DM6467 EVM board and we achieved approximately 46% cycle reduction, compared with that of FFmpeg.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.257-258
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• 2006

In this paper, we address several issues in the real time implementation of MPEG-1 Layer II decoder on a fixed-point digital signal processor (DSP), especially TMS320C6416. There is a trade-off between processing speed and the size of program/data memory for the optimal implementation. In a view of the speed optimization, we first convert the floating point operations into fixed point ones with little degradation in audio quality, and then the look-up tables used for the inverse quantization of the audio codec are forced to be located into the internal memory of the DSP. And then, window functions and filter coefficients in the decoder are precalculated and stored as constant, which makes the decoder faster even larger memory size is required. It is shown from the real-time experiments that the fixed-point implementation enables us to make the decoder with a sampling rate of 48 kHz operate with 3 times faster than real-time on TMS320C6416 at a clock rate of 600 MHz.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.1956-1959
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• 2003

본 논문에서는 HDTV(High Definition TV) 방송수신을 위한 DSP(Digital Signal Processor)기반의 HD급 비디오/오디오 디코더 시스템을 개발하고 그 성능을 확인하였다. DSP 플랫폼은 TI(Texas Instrument)사의 TMS320C6415를 대상으로 하였으며 TI의 DSP RTOS인 DSP/ BIOS를 이용하여 방송스트림인 TS(Transport Stream)을 분리하기 위한 TS Demuxer, MPEG-2 비디오 디코더 및 AC-3 오디오디코더 알고리즘을 통합하였으며, 각각의 알고리즘은 대상 DSP플랫폼인 TMS320C64x에 맞게 고정소수점 구조화 및 최적화를 실시하였다. 테스트를 위한 시스템은 스트리밍을 위한 호스트 PC와 PCI(Peripheral Component Interconnect)버스를 통해 연결된 DSP보드로 구성하였으며 실제 HDTV당송용 스트림과 SD(Standard Definition)급 스트림을 이용하여 성능을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.10C
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• pp.983-991
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• 2009

While H.264/AVC is in wide use for multimedia applications such as DMB and IPTV service, we have limited usage cases for embedded real-time applications due to its high computational demand. The paper provides judicious guide line for optimization method selection, by presenting the detailed experiments data through the development process of a real time H.264 software encoder on embedded DSP. The experimental analysis includes an intensive profiling analysis, fast algorithm application, optimal memory assignment, and intrinsic-based instruction selection. We have realized a real-time software that encodes CIF resolution videos 15 fps on TMS320DM64x processors.

• 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.

• Proceedings of the KIEE Conference
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• 2003.07d
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• pp.2681-2683
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• 2003

본 논문에서는 HDTV(High Definition TV) 방송수신을 위한 DSP기반의 HD급 비디오/오디오 디코더 시스템을 개발하고 그 성능을 확인하였다. DSP 플랫폼은 TI사의 TMS320C6415를 대상으로 하였으며 TI의 DSP RTOS인 DSP/BIOS를 이용하여 방송스트림인 TS (Transport Stream)을 분리하기 위한 TS Demuxer, MPEC-2 비디오 디코더 및 AC-3 오디오 디코더 알고리즘을 통합하였으며, 각각의 알고리즘은 대상 DSP 플랫폼인 TMS320C64x에 맞게 고정소수점 구조화 및 최적화를 실시하였다. 테스트를 위한 시스템은 스트리밍을 위한 호스트 PC와 PCI버스를 통해 연결된 DSP보드로 구성하였으며 실제 HDTV방송용 스트림과 SD급 스트림을 이용하여 성능을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.17-23
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• 2004

The mobile radio channel can be simulated as a complex-valued random process with narrow-band spectrum. This paper describes a real-time implementation of that process using a INS320C6414 digital signal processor and XC2VP30 Virtex FPGA. The simulator presented here is not only a comprehensive model of the flat fading but also frequency selective fading mobile channel conditions. To replicate the statistical characteristics of the multipath fading environment with the minimum computational burden, multi-rate techniques are employed to resolve practical problems such as variable sampling rate. The simulator produces accurate and consistent results due to digital implementation. It is very flexible and simple to program for various field conditions in mobile communications with a graphical user interface.