• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1C
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• pp.84-89
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• 2002

In this paper the Viterbi decoder which is used for TCM decoding in wireless modem system under transmission of audio data for the high quality sound is designed by VHDL and implemented by FPGA. After making short explanation about TCM encoding and decoding. I show the effect of channel in computer by using encoder and decoder implemented in FOGA and the bit error rate according to change rate of ($E_b/N_0$).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8B
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• pp.1412-1420
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• 1999

In order to enhance multi-channel audio signals, Dolby Pro Logic is widely used especially for the Hi-Fi audio system, since it can provide highly stereophonic effects and a nice separation of multi-channel sound. This paper describes an implementation of Dolby Pro Logic decoder with ARM-7 core. The code is modified for the fixed point operation and optimized. For the verification of the code, the operation time and the precision are estimated thoroughly. As a result, it is verified that Dolby Pro Logic decoder can be implemented with ARM-7 core operating at 54 MHz.

• 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.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1567-1570
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• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.875-876
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• 2008

Parametric stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus small amount of parametric overhead to describe the stereo image. The stereo properties are analyzed, encoded, and reinstated in a decoder according to spatial psycho-acoustical principles. However, coding of multichannel audio signal using parametric stereo still requires considerable bit-rate. In this paper, enhanced parametric stereo coding for multichannel audio signal is proposed.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.538-541
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• 1999

MPEG-2 AAC(Advanced Audio Coding) is the most advanced coding scheme available for high quality audio coding. This MPEG-2 AAC audio Standard allows for ITU-R ‘indistinguishable’ quality according to at data rates of 320 kb/s for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer 3, you get the same quality at 70% of the bitrate. This paper suggest optimization method for MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform) in Encoder and Decoder for AAC.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.178-181
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• 2000

In this paper, we present the implementation of filterbank for MPEG-2 Advanced Audio Coding (AAC) decoder with VHDL. The filterbank of AAC employs a technique called time-domain aliasing cancellation (TDAC). In order to make the algorithm more efficiently, we decompose and reorganize the filterbank algorithm lot the high speed decoding process and lower computational cost. And we make this filterbank algorithm to be used with other modules of AAC decoder in parallel processing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.874-882
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• 2004

MPEG Audio compression algorithm is the international standard for the digital compression of high quality audio using mechanism of the perceptual coding based on psychoacoustic masking. It is necessary to discuss the constraints on designing of common filter banks for MPEG-2 BC and MPEG-2 AAC decoder system, which is not Down yet, mapping audio signals from the time domain into the frequency domain. In this paper, we present an architecture of common synthesis filter whcih can be used for MPEG-2 BC and MPEG-2 AAC decoder using recursive structure. The proposed algorithm is based on recursive architecture that effectively performs common compulsion.

• The Magazine of the IEIE
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• v.28 no.4
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• pp.103-104
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• 2001