• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2060-2063
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• 2003

Terrestrial DMB(Digital Multimedia Broad-casting) system that is now under standardization in Korea offers multimedia broadcasting services at mobile environment and is based on Eureka-147 DAB(Digital Audio Broadcasting) for transmission method. Also DMB provides the error protection method of convolution coding. In this paper, we study on the effective error resilience coding of MPEG-4 video stream over DMB system. In our algorithm, the first, we partition the MPEG-4 data using the MPEG-4 data partitioning method, and then controls the convolution coding rate according to the importance of the partitioned data. From our simulation result, we show that our algorithm is proper for terrestrial DMB services.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.610-617
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• 2009

An overview of various modes and tools of MPEG Surround is provided Because the binaural mode of MPEG Surround supports the virtual 5.1-channel playback based on HRTFs, it can be played via headphones and earphones for portable audio devices. MPEG Surround also supports the enhanced matrix mode which converts stereo signals to 5.1-channel signals without side information, the 3D stereo mode which deals with 3D-coded signals, the low power version which greatly reduces the computational load in the decoding process. Besides, MPEG Surround provides the arbitrary downmix gains (ADGs) tool which is applied to artistic downmix signals, the matrix compatibility tool which is applied to downmix signals by conventional matrix-based methods, the residual coding tool -which can be used at high bit rates, and the GES tool which is applied to specific sound such as applause. The listening test results by various companies and organizations are also presented for important modes and tools.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• ETRI Journal
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• v.29 no.1
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• pp.103-106
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• 2007

Pilot-based coding (PBC), which is used for lossless bit rate reduction of audio coding, has been recently proposed for MPEG Surround. We propose extended PBC for further lossless bit rate reduction of MPEG Surround. Extended PBC selects the number of pilots depending on the parameter band number and the type of spatial parameter. It then encodes the pilots and the relevant difference data. Experiments show that extended PBC is more effective than the original PBC, especially for high bit rate modes, with a negligible complexity increase on the decoder side.

• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.4 s.334
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• pp.61-68
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• 2005

A chip, which can fast encoder the audio data to AAC (Advanced Audio Coding) LC(Low Complexity) that is MPEG-2 audio standard, has been designed on the basis of a 32 bits DSP core and fabricated with 0.25um CMOS technology. At first, the various optimization methods for implementing the algerian are devised to reduce the memory size and calculation cycles. FFT(Fast Fourier Transform) hardware block is added to the DSP core to get the more reduction of the calculation cycles. The chips has the size of $7.20\times7.20 mm^2$ and about 830,000 equivalent gates, can carry out AAC encoding under 70MIPS(Million Instructions per Second).

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.708-711
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• 1999

MDCT( Modified Discrete Cosine Transform ) is one of the most compute-intensive operations in the MPEG audio coding standard. In this paper a fast algorithm to perform MDCT operation is presented. The algorithm presented in the MPEG audio coding standard requires (N/2) $\times$ N multiplications and (N/2) $\times$ (N-1) additions to generate the result, but the algorithm presented in this paper requires (N/2) $\times$ (N/2) multiplications and (N/2) $\times$ (N/2) additions to perform the same task. In this algorithm N should be multiple of 4. The algorithm was implemented using ARM processor and the processing time comparison between the original algorithm and the fast algorithm is presented.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.393-398
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• 2002

The conventional bit allocation scheme for MPEG-1 audio encoding searches the subband with minimum MNR(mask-to-noise ratio) repetitively until its operation is completed, which occupies most of its total computational complexity. In this paper, as a computationally efficient approximation of it, we propose a new bit allocation scheme with a simple subband search and compare it with the existing schemes[1][2] in terms of the computational complexity and sound quality. For the performance comparison, we used the pop music signal contained in SQAM(sound quality assess material) CD from EBU. Simulation results show that the computational complexity of the proposed method is about 42% of that of the existing one in [1] and the sound quality difference in terms of MNR between the two schemes is within the 0.2 ㏈, for the case of using the layer II at the bit rate of 128 kbps.