• Journal of Intelligence and Information Systems
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• v.15 no.4
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• pp.201-212
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• 2009

Even though the spread spectrum method is known as most robust algorithm to general attacks, it has a drawback to the time axis attack. In this paper, I proposed a robust audio copyright protection algorithm which is robust to the time axis attack and has advantages of the spread spectrum method. Time axis attack includes the audio length variation attack with same pitch and the audio frequency variation attack. In order to detect the embedded watermark by the spread spectrum method, the detection algorithm should know the exact rate of the time axis attack. Even if there is a method to know the rate, it needs heavy computational resource and it is not possible to implement. In this paper, solving this problem, the audio signal is transformed into time-invariant domain, and the spread spectrum watermark is embedded into the audio in the domain. Therefore the proposed algorithm has the advantages of the spread spectrum method and it is also robust to the time axis attack. The time-invariant domain process is that the audio is arranged by log scale time axis, and then, the Fourier transform is taken to the audio in the log scale time axis. As a result, the algorithm can get the time-invariant watermark signal.

• Journal of Korea Multimedia Society
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• v.19 no.5
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• pp.826-836
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• 2016

Mastering a musical instrument for an unskilled beginning learner is not an easy task. It requires playing every note correctly and maintaining the tempo accurately. Any music comes in two forms, a music score and it rendition into an audio music. The proposed method of assisting beginning music players in both aspects employs two popular pattern recognition methods for audio-visual analysis; they are support vector machine (SVM) for music score recognition and hidden Markov model (HMM) for audio music performance tracking. With proper synchronization of the two results, the proposed music learning assistant system can give useful feedback to self-training beginners.

• ETRI Journal
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• v.40 no.1
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• pp.133-145
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• 2018

Robustness, payload, and imperceptibility of audio watermarking algorithms are contradictory design issues with high-level security of the watermark. In this study, the major issue in achieving high payload along with adequate robustness against challenging signal-processing attacks is addressed. Moreover, a security code has been strategically used for secure transmission of data, providing tamper detection at the receiver end. The high watermark payload in this work has been achieved by using the complementary features of third-level detailed coefficients of discrete wavelet transform where the human auditory system is not sensitive to alterations in the audio signal. To counter the watermark loss under challenging attacks at high payload, Daubechies wavelets that have an orthogonal property and provide smoother frequencies have been used, which can protect the data from loss under signal-processing attacks. Experimental results indicate that the proposed algorithm has demonstrated adequate robustness against signal processing attacks at 4,884.1 bps. Among the evaluators, 87% have rated the proposed algorithm to be remarkable in terms of transparency.

• Proceedings of the Korea Information Processing Society Conference
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• 2024.05a
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• pp.692-695
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• 2024

With the recent advancements in artificial intelligence (AI), the performance of deep learning-based audio deepfake technology has significantly improved. This technology has been exploited for criminal activities, leading to various cases of victimization. To prevent such illicit outcomes, this paper proposes a deep learning-based audio deepfake detection model. In this study, we propose CoNSIST, an improved audio deepfake detection model, which incorporates three additional components into the graph-based end-to-end model AASIST: (i) Squeeze and Excitation, (ii) Positional Encoding, and (iii) Reformulated HS-GAL, This incorporation is expected to enable more effective feature extraction, elimination of unnecessary operations, and consideration of more diverse information, thereby improving the performance of the original AASIST. The results of multiple experiments indicate that CoNSIST has enhanced the performance of audio deepfake detection compared to existing models.

• Journal of the Semiconductor & Display Technology
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• v.10 no.2
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• pp.121-124
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• 2011

Augmented Reality integrates virtual objects onto a real world so that it extends the human's sensibility of real-world. an Augmented Reality technology combines real and virtual object in a real environment, and runs interactive in real time, and is regarded as an emerging technology in a large part of the future of information technology. So the benefits for the various businesses are estimated to be very high. In this paper, combine ARToolkit with OpenAL we can provide audio to users. These proposed methodologies will contribute to a better immersive realization of the conventional Augmented Reality system.

• Journal of the Institute of Convergence Signal Processing
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• v.15 no.1
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• pp.1-8
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• 2014

Digital watermarking techniques are attracting attention as a proper solution to protect copyright for multimedia data. This paper proposes a new audio watermarking method based on Discrete Cosine Transformation (DCT) and Discrete Wavelet Transformation (DWT) for copyright protection. In our proposed watermarking method, the original audio is transformed into DCT domain and divided into two parts. Synchronization code is applied on the signal in first part and 2 levels DWT domain is applied on the signal in second part. The absolute value of DWT coefficient is divided into arbitrary number of segments and calculates the energy of each segment and middle peak. Watermarks are then embedded into each middle peak. Watermarks are extracted by performing the inverse operation of watermark embedding process. Experimental results show that the hidden watermark data is robust to re-sampling, low-pass filtering, re-quantization, MP3 compression, cropping, echo addition, delay, and pitch shifting, amplitude change. Performance analysis of the proposed scheme shows low error probability rates.

• Journal of Korean Society for Quality Management
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• v.44 no.4
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• pp.951-963
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• 2016

Purpose: This study was conducted to analyze the potential customer's satisfaction for the concepts of smart audio features and utilize the results when developing the customer-oriented products. Methods: 16 different features were derived via the market research and professionals' interviews. The most satisfactory features were selected through "Kano model", the relative importance of Customer Satisfaction Coefficient, and respondents' preferences from 339 valid survey answers. Results: 15 out of the 16 features were categorized as attractive attribute. "'User Recognizing' and 'Strengthen Linking' groups", such as Auto connection with Smart-phone music player, Synchronization of TV & Audio, and Volume control situational awareness, were shown to provide higher satisfactions to those potential customers. On the other hand, Group 'Integrating Function', such as Aromatherapy and Auto lighting reaction, was shown to be relatively least preferred features. Conclusion: This study enabled which features could lead to the customer satisfaction. Nevertheless, it still requires extensive analyses in different countries and diverse cultures to target the global market. The audio product planners and R&D professionals are expected to learn useful information from such studies.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1567-1570
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• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• Journal of Computing Science and Engineering
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• v.6 no.1
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• pp.40-50
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• 2012

Increase in the number of older people due to demographic changes poses great challenges to the social healthcare systems both in the Western and as well as in the Eastern countries. Support for older people by formal care givers leads to enormous temporal and personal efforts. Therefore, one of the most important goals is to increase the efficiency and effectiveness of today's care. This can be achieved by the use of assistive technologies. These technologies are able to increase the safety of patients or to reduce the time needed for tasks that do not relate to direct interaction between the care giver and the patient. Motivated by this goal, this contribution focuses on applications of acoustic technologies to support users and care givers in ambient assisted living (AAL) scenarios. Acoustic sensors are small, unobtrusive and can be added to already existing care or living environments easily. The information gathered by the acoustic sensors can be analyzed to calculate the position of the user by localization and the context by detection and classification of acoustic events in the captured acoustic signal. By doing this, possibly dangerous situations like falls, screams or an increased amount of coughs can be detected and appropriate actions can be initialized by an intelligent autonomous system for the acoustic monitoring of older persons. The proposed system is able to reduce the false alarm rate compared to other existing and commercially available approaches that basically rely only on the acoustic level. This is due to the fact that it explicitly distinguishes between the various acoustic events and provides information on the type of emergency that has taken place. Furthermore, the position of the acoustic event can be determined as contextual information by the system that uses only the acoustic signal. By this, the position of the user is known even if she or he does not wear a localization device such as a radio-frequency identification (RFID) tag.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.210-216
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• 2013

It is hard to listen to the music clearly in the presence of background noise. In this paper, a method that modifies the audio signal automatically to enhance the audio listening experience in adverse environment is proposed. Specifically, the method that amplifies the audio signal so that the perceived loudness of audio signal in each band becomes similar to that of the noiseless signal. The loudness perception model proposed by Moore et. al is utilized. Extending the previous work that is applied to speech reinforcement, the full band signal sampled at 48kHz is manipulated based on the loudness restoration principle. Moreover, based on the observation that the audio clarity is compromised even with loudness restored signal, a modification that intentionally boosts high frequency loudness more than lower band is also proposed. Experimental results showed that the proposed algorithm can enhance the audio listening experience in adverse environment.