• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.269-275
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• 2007

Pervasive Multi-channel audio systems are being realized due to advances in digital technology. This paper proposes an efficient system that serially connects individual speakers with bidirectional digital communication capability by means of SoC design. In particular, each speaker can identify the bit stream assigned to the speaker and convert it into analog audio. Furthermore, the speaker can self-diagnose the speaker functionality by utilizing the designed capability to measure frequencies of various square wave test signals. The proposed system running on 200MHz clock yielded restoration of analog output signal with latency of only $500{\mu}s$ compared to directly driving the speakers in a traditional way.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.199-206
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• 2013

Recently, the diverse needs of the drivers for in-vehicle infotainment systems are increasing rapidly. As a result, the infotainment systems are equipped with more convenient and human-friendly high-tech features. In this paper, we designed and implemented in-vehicle multimedia infotainment system based on embedded system that was applied various multimedia to in-vehicles. The proposed system can support independent display on each screen for the multi-channel multimedia source based on one processor(1 CPU). Therefore, our system can be reduced costs compared to other systems. This system not only displays the video and audio data in storage devices but also displays CAM, T-DMB, and DVB-T multimedia contents which are supplied in real-time services. Also, our system could multi-screen displays multimedia data in smart phone using Wi-Fi. We expect that in-vehicle infotainment systems like AVN(Audio video navigation) and RSE(Rear Seat Entertainment) could be used in various applications and reduced costs.

• ETRI Journal
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• v.38 no.5
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• pp.847-857
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• 2016

Broadcasting services such as multi/single channel HDTV and 3DTV/2DTV use a multi-channel encoder that changes the bitrate and composition of the video service depending on the time. However, this type of multi-channel encoder could cause a longer latency owing to the variable bitrate and relatively bigger size of the buffers, which results in the same delay as in 3DTV even for a conventional DTV service. On the other hand, systems built based on separate encoders, each of which is optimized for the target service, might not have such latency problems. Nevertheless, there might be a distortion problem in the image and sound at the time of a switchover between two encoders with different output bitrates and group of picture structures. This paper proposes a system that can realize a seamless video service conversion using two different video encoders optimized for each video service. An overall functional description of the video service change control server, which is a main control block for the proposed system, is also provided. The experiment results confirm the seamless switchover and reduced broadcasting latency of DTV services compared with a broadcasting system composed of a multi-channel encoder system.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1215-1218
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• 1998

본 논문은 Digital Audio Compression(AC-3) Standard 인 A-52를 기반으로 하였으며 Borland C++3.1 Compiler를 사용하여 AC-3 Decoding Algorithm 구현하였다. Input Stream은 DVD VOB File에서 AC-3 Stream만을 분리하여 사용하며 최종 출력은 16 Bit PCM File이다. AC-3의 Frame구조는 Synchronization Information, Bit Stream Information, Audio Block, Auxiliary Data, Error Check로 구성된다. Aduio Block 은 모두 6개의 Block으로 나뉘어져 있다. BSI와 Side Information을 참조하여 Exponent를 추출하여 Exponent Strategy에 따라 Exponent를 복원한다. 복원된 Exponent 정보를 이용하여 Bit Allocation을 수행하여 각각의 Mantissa에 할당된 Bit수를 계산하고 Stream으로부터 Mantissa를 추출한다. Coupling Parameter를 참조하ㅕ Coupling Channel을 Original Channel로 복원시킨다. Stereo Mode에 대해서는 Rematrixing을 수행한다. Dynamic Range는 Mantissa와 Exponent의 Magnitude를 바꾸는 것으로 선택적으로 사용할 수 있다. Mantissa와 Exponent를 결합하여 Floating Point coefficient로 만든 후 Inverse Transform을 수행하면 PCM Data를 얻을 수 있다. PC에서 듣기 위해서는 Multi Channel을 Stereo나 Mono로 Downmix를 수행한다. 이렇게 만들어진 PCM data는 PCM Data를 재생하는 프로그램으로 재생할 수 있다.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.8 no.1
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• pp.75-81
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• 2008

This paper presents a top-down approach to implement an MPEG-2 audio encoder in VLSI. As the algorithm of an MPEG-2 audio encoder is heavy-weighted and heterogeneous(to be mixture of several strategies), the encoder design process is undertaken carefully from the algorithmic level to the architectural level. Firstly, the encoding algorithm is analyzed and divided into sub-algorithms, called tasks, and the tasks are partitioned in the way of reusing the same designs. Secondly, the partitioned tasks are scheduled and synthesized to make the most efficient use of time and space. In the end, a real-time 5 channel MPEG-2 audio encoder is designed which is a heterogeneous multiprocessor system; two hardwired logic blocks and one specialized DSP processor.

• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.2007-2008
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• 2011

In this paper, we implemented active noise control systems for active barriers with attenuation of road noise. And then we constructed by multi-channel audio systems and DAQ part, high performance DSP H/W. Active noise control firmware program was implemented for multi-channel off-line/on-line estimation methods for secondary path transfer functions and FIR/IIR filter structure main noise control algorithms. To evaluate performance of implemented systems, active barriers test bed was implemented and several experiments was performed by various noise cases.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

• The Journal of the Acoustical Society of Korea
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• v.32 no.4
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• pp.346-353
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• 2013

This paper deals with the implementation of a high-quality audio system for karaoke. For improving the key/tempo changes performance, we separated the audio into many musical instrument channels. By separating musical instrument channels, high-quality key/tempo changes can be achieved and we confirmed this using the cross-correlation distribution and the MOS evaluation. The improved audio system was implemented using the TMS320C6747 DSP with fixed/floating-point operations. The implemented audio system can perform the multi-channel WMA decoding, the MP3 encoding/decoding, the wav playing, the EQ, and the key/tempo changes in real time. The WMA channels used for processing the separated instrument channels. The audio system includs the MP3 encoding/decoding function for playing and recording and the wav channel for the effect sound.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.3
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• pp.59-64
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• 2023

Fire and disaster broadcasting systems are divided into analog, digital, and network-based digital public address systems, and important specifications in network-based digital public address systems are low-latency audio, high sampling rate, and multi-channel input and output. In the past, it has been widely used to the AoE method for distinguishing based on the MAC address of the data link layer. However, this method has a problem of increasing complexity and cost. This proposal is an AoIP/UDP method, which allows communication to be easily distinguished by IP address without the need for a separate redundant network, so that the network can be freely used and configured, and cost can be reduced by reducing complexity. After implementing the AoIP/UDP method, the experimental results showed that the cost was improved with the equivalent performance with 2.66ms latency.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.63-68
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• 2016

Audio mixer is an electronic device which performs a mixing of multiple audio signals. Digital mixer having various functions and scalability is spreaded thanks to advanced DSP and IT technology. However, digital mixer is more vulnerable to stability comparing to conventional analog mixer in the digital error or software error sense because its control is executed by SW. To solve this problem, in this paper, we propose a multi-channel digital analog hybrid mixer scheme, digital mixer error detection mechanism and malfunctioning switching technique. Also we develop the audio mixer having digital-analog hybrid structure. By simulation, we can sense the error of digital mixer except power loss in a 120ms, change into analog mixer mode automatically and provide continuous broadcasting function without mixer function loss.