• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.8
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• pp.1450-1455
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• 2007

Recently DSP chips are widely used in high quality car audio to achieve high quality sound and to process various sound sources, e.g. navigation, cell phones. In this paper, we developed a set of design tools useful for developing high quality car audio systems using Philips' SAF7730 car audio chips. The tool is consist of the GUI(Graphic User Interface) program running on the Windows operating system and the interface board which performs data conversion between RS232C and I2C protocols. The developed system has been successfully applied to commecial car audio design.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.283-286
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• 2004

Due to advances in electronics very high quality audio reproduction is today possible. But the listening environment causes deviation of the audio system from the expected behavior. Firstly the listening Room significantly changes the audio signal frequencies and their phase. Secondly the position of the user in the room affects the perceived sound. With existing DSP technology it is possible to adequately correct these effects. In our work we developed a room correction system, correcting up to 7.1 channels using dual Motorola 56367 fixed point DSP's, implementing position dependent room effects measurement, real time compensation filter design and equalization filtering procedures.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.5
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• pp.112-117
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• 2007

This paper presents specialized DSP instructions and their hardware architecture for audio coding algorithms, such as the MPEG-2/4 Advanced Audio Coding(AAC), Dolby AC-3, MPEG-2 Backward Compatible(BC), etc. The proposed architecture is specially designed and optimized for the MDCT/IMDCT(Inverse Modified Discrete Cosine Transform), and Huffman decoding of the AAC decoding algorithm. Performance comparisons show a significant improvement compared with TMS320C62x and ASDSP21060 for the MDCT/IMDCT computation. In addition, the dedicated Huffman decoding accelerator performs decoding and preparing operand in only one cycle. The proposed DPU(Data Processing Unit) consists of 107,860 gates and achieves 150 MIPS.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• 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 KIPS Transactions:PartA
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• v.13A no.7 s.104
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• pp.589-598
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• 2006

In this paper, we describe the development and optimization of an embedded Biuetooth solution on an SoC platform for real-time audio streaming over a Bluetooth wireless link. The solution includes embedded Bluetooth protocol stack and profile simplemented on a virtual operating system for portability, and other optimization techniques to fully exploit the benefits of multimedia-oriented SoC. The optimization techniques implemented in this paper are memory access minimization by using on-chip scratch pad memory, codec library optimization with DSP and parallel memory access instruction set, and dynamic audio quality adjustment regarding current wireless link status. Experimental results show that the optimized solution presented in this paper can support high-qualify audio streaming without the support of external memory.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.63-67
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• 1997

The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1433-1442
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• 1998

The thesis presents a dual audio decoder which can decode both AC-3 and MPEG-2 bitstream. The MPEG-2 synthesis processi s optimized via FFT to establish the common data path with AC-'3s. A dual audio decoder consists of a DSP core which performs the control-intensive part of each algorithm and a common synthesis filter which perfomrs the computation-intensive part. All the components of the dual audio decoder have been described in VHDL and simulated with a SYNOPSYS tool. The software modeling of the DSP core was used for functional validation. After being synthesized using 0.6 .mu.m-3ML technology standard cell, the dual audio decoder was simulated at gate-level with a COMPASS tool for hardware validation.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.4
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• pp.477-482
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• 2007

This paper presents an audio decoder architecture which can decode AC-3 and MPEG-2 audio bit-streams efficiently. MPEG-2 synthesis filtering is modified by the 32-point FFT to share the common data path with the AC-3's. A programmable Audio DSP core and a hardwired common synthesis tilter are incorporated for effective decoding of two different formats.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2E
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• pp.91-97
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• 1995

This paper describes the implementation of a multi-channel audio codec for HETV. This codec has the features of the 3/2-stereo plus low frequency enhancement, downward compatibility with the smaller number of channels, backward compatibility with the existing 2/0-stereo system(MPEG-1 audio), and multilingual capability. The encoder of this codec consists of 6-channel analog audio input part with the sampling rate of 48 kHz, 4-channel digital audio input part and three TMS320C40 /DSPs. The encoder implements multi-channel audio compression using a human perceptual psychoacoustic model, and has the bit rate reduction to 384 kbit/s without impairment of subjective quality. The decoder consists of 6-channel analog audio output part, 4-channel digital audio output part, and two TMS320C40 DSPs for a decoding procedure. The decoder analyzes the bit stream received with bit rate of 384 kbit/s from the encoder and reproduces the multi-channel audio signals for analog and digital outputs. The multi-processing of this audio codec using multiple DSPs is ensured by high speed transfer of date between DSPs through coordinating communication port activities with DMA coprocessors. Finally, some technical considerations are suggested to realize the problem of real-time operation, which are found out through the implementation of this codec using the MPEG-2 layer II sudio coding algorithm and the use of the hardware architecture with commercial multiple DSPs.