• Proceedings of the KSPS conference
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• 2006.11a
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• pp.131-134
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• 2006

Quality of narrowband speech $(0{\sim}4kHz)$ can be enhanced by the bandwidth expansion technique, by which the high- band components are estimated. This paper proposes the bandwidth expansion method using the spline codebook based spectral folding. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured as the objective measurement In addition, the MOS (Mean Opinion Score) and the preference tests are performed as the subjective measurement. The results show our proposed method outperforms the existing spline based one.

• Journal of Information Technology and Architecture
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• v.11 no.2
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• pp.225-233
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• 2014

In this paper an architecture for 3D audio core algorithm evaluation database system. Due to increase of 3D audio system through multimedia device, an evaluation system is required for evaluating the 3D core algorithms for developing 3D audio system. Conventional evaluation systems have some problems. Researchers have to learn usage of evaluation system, in addition it is inefficient to use and search audio sources because audio sources are not indexed in general. To solve these problems, we design the architecture of 3D audio core algorithm evaluation database system enabling to automatically evaluate core algorithms using database management system. Also we define XML metadata scheme for information of saved audio source in database. This approach allows improving efficiency of search audio source and use of audio database.

• Proceedings of the KSPS conference
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• 2006.11a
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• pp.113-116
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• 2006

In this paper, we first implement an audio playback system for virtual reality by providing 3D audio effects to listeners. In general, such a 3D audio playback system utilizes a sound localization technique using head related transfer function (HRTF) to generate 3D audio effect. However, the 3D audio effect is degraded due to the crosstalk in the stereo loudspeaker environment. To enhance the 3D sound effect, we implement the crosstalk cancellation technique proposed by Atal and Schroeder and apply it to the 3D audio system.

• IEIE Transactions on Smart Processing and Computing
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• v.2 no.1
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2101-2102
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• 2006

In this paper, a new digital audio watermarking algorithm is presented. The proposed algorithm embeds watermark into audio signal based on human auditory system (HAS). This algorithm is a blind audio watermarking method, which does not require any prior information during watermark extraction process. This algorithm finds watermarking position using time-domain masking effect. First we insert the watermark into wavelet domain, and then we use a back-propagation neural network (BPN) to learn the characteristics of relationship between the watermark and the watermarked audio. Due to the teaming and adaptive capabilities of the BPN, the false recovery of the watermark can be greatly reduced by the trained BPN. Experimental results show that the proposed method has good inaudibility and high robustness to common audio processing attacks.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.662-675
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• 2018

Compression history detection plays an important role in digital multimedia forensics. Most existing works, however, mainly focus on digital image and video. Additionally, the existed audio compression detection algorithms aim to detect the trace of double compression. In real forgery scenario, multiple compression is more likely to happen. In this paper, we proposed a detection algorithm to reveal the compression history for MP3 audio. The statistics of the scale factor and Huffman table index which are the parameters of MP3 codec have been extracted as the detecting features. The experimental results have shown that the proposed method can effectively identify whether the testing audio has been previously treated with single/double/triple compression.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.70-75
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• 2007

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

• 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㏈.

• 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.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.631-635
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• 2004

The rapid development of digital media and communication network urgently brings about the need of data certification technology to protect IPR (Intellectual property right). This paper proposed a new watermarking method for embedding contents owner's audio signal in order to protect color image IPR. Since this method evolves the existing static model and embeds audio signal of big data, it has the advantage of restoring signal transformed due to attacks. Three basic stages of watermarking include: 1) Encode analogue ID owner's audio signal using PCM and create new 3D audio watermark; 2) Interleave 3D audio watermark by linear bit expansion and 3) Transform Y signal of color image into wavelet and embed interleaved audio watermark in the low frequency band on the transform domain. The results demonstrated that the audio signal embedding in color image proposed in this paper enhanced robustness against lossy JPEG compression, standard image compression and image cropping and rotation which remove a part of image.