• The KIPS Transactions:PartB
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• v.19B no.1
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• pp.9-18
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• 2012

Many audio watermarking researches which have been adapted to authenticate contents can not recover the original media after watermark removal. Therefore, reversible watermarking can be regarded as an effective method to ensure the integrity of audio data in the applications requiring high-confidential audio contents. Reversible watermarking inserts watermark into digital media in such a way that perceptual transparency is preserved, which enables the restoration of the original media from the watermarked one without any loss of media quality. This paper presents a new interval-based audio integrity authentication algorithm which can detect malicious tampering. To provide complete reversibility, we used differential histogram-based reversible watermarking. To authenticate audio in parts, not the entire audio at once, the proposed algorithm processes audio by dividing into intervals and the confirmation of the authentication is carried out in each interval. Through experiments using multiple kinds of test data, we prove that the presented algorithm provides over 99% authenticating rate, complete reversibility, and higher perceptual quality, while maintaining the induced-distortion low.

• Journal of the Korean Society for information Management
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• v.15 no.1
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• pp.149-164
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• 1998

The study aims to construct a system which can provide multimedia materials, specifically, digitalized video and audio materials on the internet. To accomplish this objective, it investigates technology on constructing a VOD/AOD system, current situations on video and audio data management in domestic and internatinal broadcasting institution and information centers. The study proposes a VOD/AOD system which can effectively manage and disseminate these materials on the internet.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.387-394
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• 2005

In this paper, we propose a new transmission technique of audio signals for the serial connection of the speakers of multiple-channel audio systems. Analog audio signals from a multi-channel audio system are converted into digital signals with signal processing steps and transferred to each speaker through a serial line. The signal processing steps contain data compression and packet generation in association with audio signal characteristics. Each speaker gets its corresponding digital audio signals from the transmitted packets and converts the signals into analog audio signals to make sounds with the speaker All the proposed functions in this paper are modeled in VHDL. implemented with FPGA chips, and tested for actual multi-channel audio systems.

• Proceedings of the KIEE Conference
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• 1987.07b
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• pp.1053-1057
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• 1987

In this paper, we have studied a communication protocol which may provide Audio/Video data transmission in real time. Auido/Video data have its own characteristics. A new transport protocol with realtime constraint has been designed and implemented which performs dynamic error control and flow control depending on the characteristics of transmitted Audio/Video data. Since the receiving data can be predicted from the previously received data using the prediction function in Auido/Video data transmission, these functions are introduced in our transport protocol that may possibly improve the speed of data transmission and give a real time response. We have tested our transport protocol and measured the performance by the simulation. We assume that our transport protocol would be used in LAN environment. Our prime purpose is to provide a reliable and real time Auido/Video data transmission service.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2262-2264
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• 1998

This paper propose a new method for the discrete signal : Discrete Wavelet Transform(DWT). This paper is a brief introduction to the DWT and applies the DWT coding for the audio data as an example. We can have a number of hint about the compression algorithm of multimedia resources and the high performance of transmission and storage.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.5 s.311
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• pp.48-56
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• 2006

In this paper, we proposed the high quality audio watermarking algorithm using MDCT/IMDCT (Modified DCT/Inverse Modified DCT) with psychoacoustic model. Generally, a digital audio watermark is embedding the frequency domain after frequency transform of the digital audio data but the digital audio quality is affected by watermarking. In our scheme, the digital audio data is spread with PN((Pseudo Noise) code and then audio watermark is embedded in MDCT processing that refers psychoacoustic model. In MDCT processing, according to the shape of filter bank output, the block switching selects a window sequence that has 256, 1,024 or 2,048 points interval for high quality audio. The author confirm that when watermark weight ${\alpha}$ is 2.5 below, the detection ratio of watermark is a satisfied to SDMI's(Secure Digital Music Initiative) recommendation 50% above and SM is $50{\sim}68dB$ range with mainly 4 kind of attacks(Compression, Cropping, FFT and Echo).

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.569-572
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• 2005

This paper proposed the design and implementation of a DSP platform for the various multimedia streaming. The DSP platform synchronizes with host PC to configure DSP and to transmit multimedia streaming through PCI. The suggested DSP platform decodes high-capacity video/audio data using the suggested high-speed FIFO, CPLD and memory interface. The buffer control techniques is proposed in other to avoid the under/over-run of the audio/video data during the audio/video decoding. For the DSP platform test, host PC transmits program stream(PS) that consists of the MPEG-2 video MP@ML and 5.1ch AC3 audio data (Coyote.mov file, half hour running time) to DSP platform. The DSP platform plays continuously back the high sound-quality audio and high-definition video at once.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.772-775
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• 2005

This paper describes a study on the digital watermarking algorithm which is used to confirm the copyright protection of digital audio data. The digital audio watermarking algorithm based on psychoacoustic model is used for the inaudibility of the watermark data. The psychoacoustic model which is a key algorithm in MP3 audio compression is analyzed by MATLAB simulation, and is applied to digital audio watermark insertion.

• 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 the Korea institute of electronic communication sciences
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• v.15 no.3
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• pp.465-472
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• 2020

Recently, various deep neural networks based methods have been proposed for audio event detection. In this study, we improved the performance of audio event detection by adopting an attention approach to a baseline CRNN. We applied context gating at the input of the baseline CRNN and added an attention layer at the output. We improved the performance of the attention based CRNN by using the audio data of strong labels in frame units as well as the data of weak labels in clip levels. In the audio event detection experiments using the audio data from the Task 4 of the DCASE 2018/2019 Challenge, we could obtain maximally a 66% relative increase in the F-score in the proposed attention based CRNN compared with the baseline CRNN.