• 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.

• Journal of Broadcast Engineering
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• v.28 no.2
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• pp.213-229
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• 2023

In this paper, we introduce a spatial sound rendering system that provides 6DoF spatial sound in real time in response to the movement of a listener located in a virtual environment. This system was implemented using MPEG-I AEP as a development environment for the CfP response of MPEG-I Immersive Audio and consists of an encoder and a renderer including a decoder. The encoder serves to offline encode metadata such as the spatial audio parameters of the virtual space scene included in EIF and the directivity information of the sound source provided in the SOFA file and deliver them to the bitstream. The renderer receives the transmitted bitstream and performs 6DoF spatial sound rendering in real time according to the position of the listener. The main spatial sound processing technologies applied to the rendering system include sound source effect and obstacle effect, and other ones for the system processing include Doppler effect, sound field effect and etc. The results of self-subjective evaluation of the developed system are introduced.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.165-168
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• 1998

제안하고 있는 CSD17C00 chip은 C&S technology에서 개발한 것으로, 음성 신호 처리를 위해 범용으로 구현되었으며, 16 bit 40 MIPS DSP group OAK DSP Core를 포함, 이에 Miscellaneous Logic, Serial Port, Host Interface, Timer, Compander 의 5가지 Peripherals 과 범용 I/O Ports 로 설계되었다. 1차적으로 CSD17C00 Chip 의 성능을 점검하였다. 그 결과, 응용 프로그램은 28MIPS의 계산속도를 갖으며, 프로그램 ROM 크기는 8.85KWords 이고, 10KWords 의 데이터 ROM 과 4KWords 데이터 RAM을 필요로 한다. CSD17C00 CHIP은 멀티미디어 통신용 VOCODER 개발을 위한 범용성을 갖추고 있으며, VOCODER 용 S/W 개발 환경 및 H/W 구조가 여타 범용 DSP에 비해편의성고 K합리성을 제공하도록 설계되어 있다. 따라서, 이를 이용한다면, 멀티 미디어 통신용 VOCODER, INTERNET PHONE CO-PROCESSOR, DIGITAL RECODER, MPEG AUDIO ENCODER & DECODER 등 다양한 제품으로의 응용이 가능할 것으로 전망된다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1999.06b
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• pp.79-86
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• 1999

본 논문에서는 ISO/IEC MPEG-2 AAC 표준을 기반으로 한 멀티채널 오디오 인코더 및 디코더(MASIC) 시스템의 실시간 구현 기술에 대해서 기술한다. MPEG-2 AAC 기술은 멀티채널 오디오 부호화 방식의 국제 표준으로써, 지금까지 개발된 멀티채널 오디오 부호화 방식중 최신의 기술이며, 압축율과 오디오 품질이 가장 우수한 것으로 알려져 있다. MASIC 시스템은 인코딩 및 디코딩 기술의 실시간 처리를 위하여 범용 DSP인 TMS320C6701을 사용하였고, 멀티채널 오디오의 고속 입력과 출력을 위한 디지털 인터페이스를 가지고 있으며, 개인용 컴퓨터와의 인터페이스를 위한 PCI 기술이 적용되어 다양한 입출력 모드를 지원하는 특징을 갖는다.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.592-602
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• 2003

This paper aims to describe the basic structure of novel object-based 3D audio broadcasting system To overcome current uni-directional audio broadcasting services, the object-based 3D audio broadcasting system is designed for providing the ability to interact with important audio objects as well as realistic 3D effects based on the MPEG-4 standard. The system is composed of 6 sub-modules. The audio input module collects the background sound object, which is recored by 3D microphone, and audio objects, which are recorded by monaural microphone or extracted through source separation method. The sound scene authoring module edits the 3D information of audio objects such as acoustical characteristics, location, directivity and etc. It also defines the final sound scene with a 3D background sound, which is intended to be delievered to a receiving terminal by producer. The encoder module encodes scene descriptors and audio objects for effective transmission. The decoder module extracts scene descriptors and audio objects from decoding received bistreams. The sound scene composition module reconstructs the 3D sound scene with scene descriptors and audio objects. The 3D sound renderer module maximizes the 3D sound effects through adapting the final sound to the listner's acoustical environments. It also receives the user's controls on audio objects and sends them to the scene composition module for changing the sound scene.

• Journal of Broadcast Engineering
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• v.24 no.3
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• pp.387-394
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• 2019

In this study, a new method of audio high-band coding based on autoencoder with side information is proposed. The proposed method operates in the MDCT domain, and improves the performance by using additional side information consisting of the previous and current low bands, which is different from the conventional autoencoder that only inputs information to be encoded. Moreover, the side information in a time-frequency domain enables the high-band coder to utilize temporal characteristics of the signal. In the proposed method, the encoder transmits a 4-dimensional latent vector computed by the autoencoder and a gain variable using 12 bits for each frame. The decoder reconstructs the high band by applying the decoded low bands in the previous and current frames and the transmitted information to the autoencoder. Subjective evaluation confirms that the proposed method provides equivalent performance to the SBR at approximately half the bit rate of the SBR.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.95-100
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• 2005

This paper proposes a DVS method which effectively reduces the power consumption of multimedia signal processor. Analytic derivations of effective range of its power saving factor are obtained with the assumption of a Gaussian distribution for the frame-based computational burden of the multimedia processor. A closed form equation of the power saving factor is derived in terms of the mean-standard deviation of the distribution. An MPEG-2 video decoder algorithm and AAC encoder algorithm are tested on ARM9 RISC processor for the experimental verification of the power saying of the proposed DVS approach. The experimental results with diverse MPEG-2 video and audio files show 50～30% power saving factor and show good agreement with those of the analytically derived values.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.37-44
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• 2000

In this paper, we described the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point TMS320C6701 DSP. We adopted a hardware structure for real-time operation. In order for software optimization, we used C- and assembly-language level optimizations for time-critical functional codes. Utilizing the internal program memory of the DSP as the program cache, the internal data memory overlap technique and DMA functionality, we could get a goal of realtime operation of HVXC codec both at 2 kbit/s and at 4 kbit/s. For an encoder at 2 kbit/s, the optimization ratio to original code is about 96 %. Finally, we got the subjective quality of MOS 2.45 at 2 kbit/s from an informal quality test.

• Journal of Digital Convergence
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• v.12 no.3
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• pp.269-275
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• 2014

The DAB system was designed to provide CD quality audio and data services for fixed, portable and mobile applications with the required BER below $10^{-4}$. However for the T-DMB system with the video service of MPEG-4 stream, BER should go down $10^{-8}$ by adding FEC blocks which consist of the Reed-Solomon (RS) encoder/decoder and convolutional interleaver/deinterleaver. In this paper we propose two types of turbo coded T-DMB system without altering the puncturing procedure and puncturing vectors defined in the standard T-DMB system for compatibility. One(Type 1) can replace the existing RS code, convolutional interleaver and RCPC code by a turbo code and the other one (Type 2) can substitute the existing RCPC code by a turbo code. Simulation results show that two new turbo coded systems are able to yield considerable performance gain after just 2 iterations. Type 2 system is better than type 1 but the amount of performance improvement is small.

• Smart Media Journal
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• v.11 no.10
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• pp.65-75
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• 2022

Deep learning is used as a creative tool that could overcome the limitations of existing analysis models and generate various types of results such as text, image, and music. In this paper, we propose a method necessary to preprocess audio data using the Niko's MIDI Pack sound source file as a data set and to generate music using Bi-LSTM. Based on the generated root note, the hidden layers are composed of multi-layers to create a new note suitable for the musical composition, and an attention mechanism is applied to the output gate of the decoder to apply the weight of the factors that affect the data input from the encoder. Setting variables such as loss function and optimization method are applied as parameters for improving the LSTM model. The proposed model is a multi-channel Bi-LSTM with attention that applies notes pitch generated from separating treble clef and bass clef, length of notes, rests, length of rests, and chords to improve the efficiency and prediction of MIDI deep learning process. The results of the learning generate a sound that matches the development of music scale distinct from noise, and we are aiming to contribute to generating a harmonistic stable music.