• 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.

• Proceedings of the IEEK Conference
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• 2000.06c
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• pp.125-128
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• 2000

In this paper, we proposed the high quality digital audio watermarking algorithm in the frequency domain. The spread spectrum technique is used to encrypted a stream of information by spreading the data as much of the frequency spectrum as possible. It's technique adapt well to data hiding in audio signal. We have used the perceptual model and MDCT/IMDCT for the high qualify digital audio watermarking. The proposed watermark algorithm preserved high quality audio data from watermark signal.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.454-457
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• 2002

In this paper, we proposed digital audio watermarking algorithm for high quality audio. Nowadays, digital watermark used to confirm to digital copyright protection, not only digital image but also digital audio is active in the digital watermarking study. In this paper, we proposed digital audio watermarking algorithm using psychoacoustics model and MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform) for the high quality audio watermark. In the proposed scheme, we used to 441KHz, 128kbps and stereo audio data for audio watermarking algorithm. Audio data is passed by MDCT; watermark can be inserted into the frequency domain with 256,1024 and 2048 interval.

• 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%.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.541-547
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• 2013

The speech quality of multi-party audio telephony between two, three or more participants is decreased by audio volume imbalance, audio volume saturation and noise level increase. To solve this issue, this paper proposes an advanced audio mixing algorithm for software-based multi-point control unit. Our approach is based on the combined voice activity detection and gain control technique that consists of a set of algorithms that classify audio signals, estimate audio volumes, adjust gain factors and mix audio signals of all channels. The proposed audio mixing algorithm is computationally efficient, delivers high-quality speech, and is suitable for use in any practical multi-party audio telephony.

• Journal of Korea Multimedia Society
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• v.8 no.6
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• pp.807-813
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• 2005

The production and authoring of digital multimedia contents are most important fields in multimedia technology. Nowadays web-based technology and related multimedia software technology are growing in the IT industry and these technologies are evolving most rapidly in our life. The technology of digital audio and video processing is utilizing rapidly to improve quality of our life, Also we are more interested in high sense and artistic feeling in the music and entertainment areas by use of three dimensional (3D) digital sound technology continuously as well as 3D digital video technology. The service field of digital audio contents is increasing rapidly through the Internet. And the society of Internet users wants the audio contents service with better quality. Recently Internet users are not satisfying the sound quality with 2 channels stereo but seeking the high quality of sound with 5,] channels such as 3D audio of the movie films. But it might be needed proper hardware equipments for the service of 3D sound to satisfy this demand. In this paper, we expand the simple 3D audio generator developed and propose a web-based music bank by the software development of 3D audio generation player in 3D sound environment with two speakers minimizing hardware equipments, Also we believe that this study would contribute greatly to digital 3D sound service of high quality for music and entertainment mania.

• Journal of KSNVE
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• v.9 no.5
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• pp.991-997
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• 1999

It is getting higher the demand on a good sound in a car. Like some other sensory qualities, achieving a good sound is still treated as a black-box followed by the individual taste of tuning engineers. This study aims to define the attributes of a good car audio sound and the assessment factors affecting the sound quality. With a variety of surveys on professional listening groups and car audio installers, 26 most frequently used vocabularies are selected as typical presentations of car audio sound quality. After repetitions of listening tests with deliberately selected test CD, pre-recorded 15 different digital car-recordings, headphones replay system, and three different listening groups, it is proved that there are at least 5 different assessment factors which significantly affect the car audio sound quality. They are high fidelity, spectral balance, tonal accuracy, sound stage and acoustic fault.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.3
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• pp.53-61
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• 2002

In this paper, the authors proposed the digital audio watermarking algorithm for a high quality audio. Today, the digital watermark is used to confirm to the digital copyright protection, not only the digital image but the digital audio study is an activeness in the digital watermarking area. Especially, the watermark insertion in the digital audio area affects deeply not only a robustness but the audio quality of the watermarked audio data. Generally, the audio watermark is inserted in the frequence domain after FFT, the quality of audio data is affected by the watermark insertion. Thus, a high quality audio to be maintained at the same time, the study related a inserting of the robustness watermark happened to a hot issue. In this paper, the authors proposed the digital audio watermarking algorithm using psychoacoustic model and MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform). In the proposed scheme, the authors experimented the stereo audio file with 44.1KHz, and 128kbps for the audio watermarking algorithm proposed. When the audio data is processed by MDCT, the watermark is able to insert into the frequence domain with 256, 1024 and 2048 interval. In case of 50㎳ RMS window, it was confirmed that the difference between the original audio data and the watermarked audio data of RMS power is 0.8㏈.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.538-541
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• 1999

MPEG-2 AAC(Advanced Audio Coding) is the most advanced coding scheme available for high quality audio coding. This MPEG-2 AAC audio Standard allows for ITU-R ‘indistinguishable’ quality according to at data rates of 320 kb/s for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer 3, you get the same quality at 70% of the bitrate. This paper suggest optimization method for MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform) in Encoder and Decoder for AAC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3C
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• pp.240-246
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• 2008

An MPEG-2/4 AAC decoder on 16-bit fixed-point processor is presented in this paper. To meet audio quality criteria, despite small word length, special design methods for 16-bit foxed-point AAC decoder were devised. This paper presents particular algorithms for 16-bit AAC decoding. We have implemented an efficient AAC decoder using the proposed algorithms. Audio contents can be replayed in the decoder without quality degradation.