• Journal of Internet Computing and Services
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• v.21 no.4
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• pp.51-57
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• 2020

Currently, High-efficient video coding (HEVC) has become the most promising video coding technology. However, the implementation of HEVC in video streaming systems is restricted by factors such as cost, design complexity, and compatibility with existing systems. While HEVC is considering deploying to various systems with different reached methods, H264/AVC can be one of the best choices for current video streaming systems. This paper presents an adaptive method for manipulating video streams using video coding on an integrated circuit (IC) designed with a private network processor. The proposed system allows to transfer multimedia data from cameras or other video sources to client. For this work, a series of video or audio packages from the video source are forwarded to the designed IC via HDMI cable, called Tx transmitter. The Tx processes input data into a real-time stream using its own protocol according to the Real-Time Transmission Protocol for both video and audio, then Tx transmits output packages to the video client though internet. The client includes hardware or software video/audio decoders to decode the received packages. Tx uses H264/AVC or HEVC video coding to encode video data, and its audio coding is PCM format. By handling the message exchanges between Tx and the client, the transmitted session can be set up quickly. Output results show that transmission's throughput can be achieved about 50 Mbps with approximately 80 msec latency.

• MALSORI
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• no.61
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• pp.75-86
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• 2007

In this paper, we propose a scalable audio coder, which has a variable bandwidth from the narrowband speech bandwidth to the audio bandwidth and also has a bit-rate from 8 to 320 kbits/s, in order to cope with the quality of service(QoS) according to the network load. First of all, the proposed scalable coder splits bandwidth of the input audio into narrowband up to around 4 kHz and above. Next, the narrowband signals are compressed by a speech coding method compatible to an existing standard speech coder such as G.729, and the other signals whose bandwidth is above the narrowband are compressed on the basis of a psychoacoustic model. It is shown from the objective quality tests using the signal-to-noise ratio(SNR) and the perceptual evaluation of audio quality(PEAQ) that the proposed scalable audio coder provides a comparable quality to the MPEG-1 Layer III (MP3) audio coder.

• ETRI Journal
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• v.28 no.5
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• pp.664-667
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• 2006

We propose an efficient audio transcoding algorithm that can convert audio streams from terrestrial digital television broadcasting service stations to those for terrestrial digital multimedia broadcasting hand-held receivers. The proposed algorithm avoids the complicated psychoacoustic analysis by calculating the scalefactors of the bit-sliced arithmetic coding encoder directly from the signal-to-noise ratio parameters of the AC-3 decoder. The bit-allocation process is also simplified by cascading the nested distortion control loop. Through subjective evaluation, it is shown that the proposed algorithm provides comparable audio quality to tandem coding but it requires much smaller complexity.

• Proceedings of the IEEK Conference
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• summer
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• pp.114-117
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• 2004

This paper proposes a novel algorithm for hybrid lossless audio coding, which employs integer discrete cosine transform. The proposed algorithm divides the input signal into frames of a proper length, decorrelates the framed data using the integer DCT and finally entropy-codes the frame data. In particular, the adaptive Golomb-Rice coding method used for the entropy coding selects an optimal option which gives the best compression efficiency. Since the proposed algorithm uses integer operations, it significantly improves the computation speed in comparison with an algorithm using real or floating-point operations. When the coding algorithm is implemented in hardware, the system complexity as well as the power consumption is remarkably reduced. Finally, because each frame is independently coded and is byte-aligned with respect to the frame header, it is convenient to move, search, and edit the coded data.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1
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• pp.18-25
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• 2008

In this paper, new lossless coding method of spectral coefficients for audio codec is proposed. Conventional lossless coder uses Huffman coding utilizing the statistical characteristics of spectral coefficients, but does not provide the high coding efficiency due to its simple structure. To solve this limitation, new lossless coding scheme with better performance is proposed that consists of bit-plane transform and run-length coding. In the proposed scheme, the spectral coefficients are first transformed by bit-plane into 1-D bit-stream with better correlative properties, which is then coded intorun-length and is finally Huffman coded. In addition, the coding performance is further increased by applying the proposed bit-plane coding selectively to each group, after the entire frequency is divided into 3 groups. The performance of proposed coding scheme is measured in terms of theoretical number of bits based on the entropy, and shows at most 6% enhancement compared to that of conventional lossless coder used in AAC audio codec.

• Journal of Internet Computing and Services
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• v.10 no.5
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• pp.33-39
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• 2009

In this paper, the Window based dual-mode audio receiver of DAB and DAB+(Digital Audio Broadcasting Plus) is designed, and audio performance of it is analyzed. DAB+ can be composed of DAB and AAC(Advanced Audio Coding) for more effective audio services in the limited channel bandwidth. In the result of thesis, the Window based receiver can simultaneously be decoded for DAB/DAB+ dual-mode audio.

• IEMEK Journal of Embedded Systems and Applications
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• v.2 no.1
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• pp.31-42
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• 2007

A transform coding scheme for the transmission of audio data in ZigBee based wireless personal area networks (WPAN) is presented in this paper. Here, wavelet transform is exploited to encode the features of audio data included mainly in the low frequency region. As a result, it is confirmed that the presented scheme recovers the original audio signals much accurately while it transmits the binary data compressed as 37.5% of the entire data generated without coding scheme. Especially, the mean-squared error between the recovered and original audio data approaches $10^{-4}$ when the signal-to-noise power ratio is sufficiently high. Hence, the presented coding scheme which exploits the wavelet transform is possibly applied for high-quality audio data transmission services in a small-scale sensor network based on ZigBee. Such a result is considered to be applicable as a basic material to update the technical specifications and develop the applications of ZigBee in WPANs.

• ETRI Journal
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• v.38 no.5
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• pp.829-837
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• 2016

In this paper, we introduce extension tools for MPEG Surround, which were recently adopted as MPEG-H 3D Audio tools by the ISO/MPEG standardization group. MPEG-H 3D Audio is a next-generation technology for representing spatial audio in an immersive manner. However, considerably large numbers of input signals can degrade the compression performance during a low bitrate operation. The proposed extension of MPEG Surround was basically designed based on the original MPEG Surround technology, where the limitations of MPEG Surround were revised by adopting a new coding structure. The proposed MPEG-H 3D Audio technologies will play a pivotal role in dramatically improving the sound quality during a lower bitrate operation.

• ETRI Journal
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• v.27 no.3
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• pp.330-333
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• 2005

The Inter-Channel Level Difference (ICLD) is a cue parameter to estimate spectral information in a binaural cue coding that has been recently in the spotlight as a multichannel audio signal compression technique. Even though the ICLD is an essential parameter, it is generally distorted by quantization. In this paper, a new modified ICLE representation method to minimize the quantization distortion is proposed by adopting a flexible determination of the reference channel and the unidirectional quantization. Our experimental result confirms that the proposed method improves the multichannel audio output quality even with the reduced bit-rate.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.157-160
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• 2004

Spatial Audio Coding 방법 중 하나인 Binaural Cue Coding 방법은 다채널 다객체 오디오 신호를 모노나 스테레오로 다운 믹스한 신호와 spatial 큐를 전송해 디코더에서 복원하는 기술로 작은 비트 율로 다채널 오디오 신호를 전송 복원해 내는 기술이다. 본 논문은 BCC 코딩 방법에서 채널 상관도를 나타내는 ICC 파라메터에 따라 spatial cue 종류를 달리함으로써 전송되는 부가정보의 비트 율을 줄이는 방법을 제안한다.