• The Journal of the Acoustical Society of Korea
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• v.16 no.3E
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• pp.56-61
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• 1997

In this paper, we describe a real-time implementation of a multi-channel auido codec system that is based on the MPEG-1 audio algorithm. The major feature of this system is that it has a flexible multi-DSP system that can be adapted for various applications with using up to four TMS320C40 DSPs. The purpose of this paper is to present the problems of the system and is to describe the optimized methods to solve the problems in the view of hardware and software. Our audio codec is composed of an encoder an a decoder system and the bit rate of bitstream is up to 384 kbps. Fast input/output interfaces, DSP overloads, and inter-DSP communications methods with high speed are considered in multi-DSP H/W. Also, to run real-time in S/W, optimizing methods of algorithm are considered. After implementation of system, the subjective assessment method, and 'triple stimulus/hidden reference/double blind' that recommended by ITU-R TG10/3 is adopted for the quality of our system. All test items except one are awarded difference grades(diffgrade) better than 1-. Form the results, multi-channel audio system can be used for HDTV service.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.109-114
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• 2007

In this paper, we Propose a pyramid VQ to quantize the transform coefficients of TCX module for the audio improvement of AMR-WB+ codec. The Proposed pyramid VQ is compared to the $RE_8$ Lattice VQ used in the AMR-WB+ standard codec. demonstrating improvement 4% and 5.7%. respectively, in Mean Squared Error (MSE) and 3.3% and 4.7%. respectively, in Perceptual Evaluation of Audio Quality (PEAQ) by 8-dimensional and 16-dimensional Pyramid VQ.

• Journal of the Korea Society of Computer and Information
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• v.14 no.8
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• pp.51-58
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• 2009

In this paper. we propose search algorithm using Low Peak Feature of audio and image color value by which similar movies can be identified. Combing through entire video files for the purpose of recognizing and retrieving matching movies requires much time and memory space. Moreover, these methods still share a critical problem of erroneously recognizing as being different matching videos that have been altered only in resolution or converted merely with a different codec. Thus we present here a similar-video-retrieval method that relies on analysis of audio patterns, whose peak features are not greatly affected by changes in the resolution or codec used and image color values. which are used for similarity comparison. The method showed a 97.7% search success rate, given a set of 2,000 video files whose audio-bit-rate had been altered or were purposefully written in a different codec.

• Journal of Broadcast Engineering
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• v.25 no.2
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• pp.233-241
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• 2020

In this paper, a new buffer control method is proposed for improving the performance of the frequency domain part of the 3D audio (3DA) core codec. For the proposed buffer control method, we first combine the 3DA RM9 with the 3GPP AAC buffer control method which includes the psychoacoustic model and rate-distortion control process with the spectral hole avoidance algorithm. Then, we revise the 3GPP buffer control method so as to achieve a faithful bit allocation to the frames with higher activity. With the MUSHRA test, we prove that the proposed buffer control method demonstrates better performance than the 3DA RM9 and 3GPP AAC.

• Journal of Broadcast Engineering
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• v.26 no.3
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• pp.327-330
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• 2021

In this paper, a new noise filling method is proposed for improving the performance of the 3D audio codec. In the new method, the core band is limited up to MAX_SFB, not up to the IGF start frequency. And the noise filling is applied to all frequency range of the IGF source patches. We conduct the MUSHRA test and find that the proposed noise filling method demonstrates better performance than the conventional method.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.43-48
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• 2000

This paper introduces MPEG-2 AAC codec system fur digital audio broadcasting. This system consists of encoder and decoder, and this system provides MPEG-2 system multiplexing and demultiplexing functions. Four DSPs are adopted fur encoder and three DSPs fur decoder. Each DSP processes system control, I/O control, and audio signal processing, multiplexing and demultiplexing. This paper also discusses about some near future estimations related to DAB system and services. And at the end of this paper describes about future development plans.

• Journal of Internet Computing and Services
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• v.17 no.4
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• pp.11-17
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• 2016

In this paper, we proposed a multi-viewer system with multiple HD cameras based AVB(Audio Video Bridge) ethernet cable using IP networking, and FPGA(Xilinx Zynq 702) for bus safety systems. This AVB (IEEE802.1BA) system can be designed for the low latency based on FPGA, and transmit real-time with HD video and audio signals in a vehicle network. The proposed multi-viewer platform can multiplex H.264 video signals from 4 wide-angle HD cameras with existed ethernet 1Gbps. and 2-wire 100Mbps cables. The design of Zynq 702 based low latency to H.264 AVC CODEC was proposed for the minimization of time-delay in the HD video transmission of car area network, too. And the performance of PSNR(Peak Signal-to-noise-ratio) was analyzed with the reference model JM for encoding and decoding results in H.264 AVC CODEC. These PSNR values can be confirmed according the theoretical and HW result from the signal of H.264 AVC CODEC based on Zynq 702 the multi-viewer with multiple cameras. As a result, proposed AVB multi-viewer platform with multiple cameras can be used for the surveillance of audio and video around a bus for the safety due to the low latency of H.264 AVC CODEC design.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.13 no.2
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• pp.176-187
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• 2002

The digital Audio Broadcasting (DAB) system adopts Coded OFDM(COFDM) for channel coding. The COFDM is a combined technique of multicarrier transmission(OFDM) and punctured convolutional coding with viterbi error correction. Because the channel coding is an important topic for OFDM systems, this paper proposes a new turbo coded OFDM system that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the frequency selective Rician fading channel and the frequency selective Rayleigh fading channel in conjunction with DAB transmission mode I suitable for the terrestrial single frequency network(SFN) broadcasting. The standard system's performance was improved with the aid of turbo codec.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3
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• pp.262-267
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• 2004

AMR-WB (Adaptive Multi Rate Wideband) speech codec, the most recent voice codec standardized by 3GPP, has the wider audio bandwidth of 50∼7000 Hz and operates on nine speech coding bit rates between 6.60 and 23.85 kbit/s. This Paper presents the real-time implementation of AMR-WB speech codec by using a 16 bit fixed-point TeakLite DSP. The implemented AMR-WB codec requires the complexity of 52.2 MIPS at 23.85 kbit/s mode and also needs the program memory of 17.9 kwords, data RAM of 11.8 kwords, and data ROM of 10.1kwords. It was verified through passing the all test vectors provided by 3GPP with maintaining bit exactness. Stable operations on the real-time testing board were also proved without any distortions and delays for the audio in/out.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1990.06a
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• pp.648-654
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• 1990