• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.812-821
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• 2009

This paper proposes a novel dynamically reconfigurable switch for various multimedia applications in hybrid NoC systems. Current NoC systems, which adopt hybrid NoC structure with fixed switch and job distribution algorithms, require designers to precisely predict the property of applications to be processed. This paper proposes a reconfigurable switch which minimizes buffer overflow in various multimedia applications running on an NoC system. To verify the performance of the proposed system, we performed experiments on various multimedia applications running on embedded systems, such as MPEG4 and MP3 decoder, GPS positioning system, and OFDM demodulator. Experimental results show that buffer overflow has been decreased by 41.8% and 29.0%, respectively, when compared with NoC systems having sub-clusters with mesh or star topology. Power usage has been increased by 2.3% compared with hybrid NoC systems using fixed switches, and chip area has been increased from -0.6% to 5.7% depending on sub-cluster topology.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.6 no.2
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• pp.95-106
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• 2007

This paper proposes an effective tempo feature extraction method for music information retrieval. The tempo information is modeled by the narrow-band temporal modulation components, which are decomposed into a modulation spectrum via joint frequency analysis. In implementation, the tempo feature is directly extracted from the modified discrete cosine transform coefficients, which is the output of partial MP3(MPEG 1 Layer 3) decoder. Then, different features are extracted from the amplitudes of modulation spectrum and applied to different music information retrieval tasks. The logarithmic scale modulation frequency coefficients are employed in automatic music emotion classification and music genre classification. The classification precision in both systems is improved significantly. The bit vectors derived from adaptive modulation spectrum is used in audio fingerprinting task That is proved to be able to achieve high robustness in this application. The experimental results in these tasks validate the effectiveness of the proposed tempo feature.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.69-72
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• 2003

This paper presents a signal processing IC for digital HDTV, which is designed to operate in bunt-in HDW or in HD-set-top Box. The chip supports de-multiplexing an ISO/IEC 13818-1 MPEG-2 TS stream. It decodes MPEG-2 MP@HL video bitstream, and provides high-quality scaled video for display on HDTV monitor. The chip consists of ARM7TDMI for TS-Demux, PCI interface, Audio interface, MPEG2 MP@HL video decoder Display processor, Graphic processor, Memory controller, Audio int3face, Smart Card interface and UART. It is fabricated using Sam sung's 0.18-um and the package of 492-pin BGA is used.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.3
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• pp.1684-1699
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• 2017

The deblocking filter (DF) reduces blocking artifacts in encoded video sequences, and thereby significantly improves the subjective and objective quality of videos. Statistics show that the DF accounts for 5-18% of the total decoding time in high-efficiency video coding. Therefore, speeding up the DF will improve codec performance, especially for the decoder. In view of the rapid development of multicore technology, we propose a parallel DF scheme based on a modified order of accessing the coding tree units (CTUs) by analyzing the data dependencies between adjacent CTUs. This enables the DF to run in parallel, providing accelerated performance and more flexibility in the degree of parallelism, as well as finer parallel granularity. We additionally solve the problems of variable privatization and thread synchronization in the parallelization of the DF. Finally, the DF module is parallelized based on the HM16.1 reference software using OpenMP technology. The acceleration performance is experimentally tested under various numbers of cores, and the results show that the proposed scheme is very effective at speeding up the DF.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.157-159
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• 2014

In this paper, we introduce SlimSRP, an ultra-low-power digital signal processor (DSP) solution for mobile audio and voice applications. So far, application processors (APs) have taken charge of all the tasks in mobile devices. However, they have suffered from short battery life problems to deal with complex usage scenarios, such as always-on voice trigger with continuous audio playback. From extensive analysis of audio and voice application characteristics, SlimSRP is designed to relive the performance and power burden of APs. It employs three-issue VLIW architecture, and the major low-power and high-performance techniques include: (1) an optimized register-file architecture friendly for constants generation, (2) a powerful instruction set to reduce the number of register file accesses and (3) a unique instruction compression scheme that contributes to saved memory size and reduced cache miss. An implementation of SlimSRP runs at up to 200MHz and the logic occupies 95K NAND2 gates in Samsung 28LPP process. The experimental results demonstrate that a MP3 decoder application with a 128kbps 44.1kHz input can run at 5.1MHz and the logic consumes only 22uW/MHz.