• ETRI Journal
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• v.27 no.5
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• pp.608-616
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• 2005

Watermarking has become a technology of choice for a broad range of multimedia copyright protection applications. This paper proposes an audio watermarking scheme that uses the modified tonal masker as an embedding carrier for imperceptible and robust audio watermarking. The method of embedding is to select one of the tonal maskers using a secret key, and to then modify the frequency signals that consist of the tonal masker without changing the sound pressure level. The modified tonal masker can be found using the same secret key without the original sound, and the embedded information can be extracted. The results show that the frequency signals are stable enough to keep embedded watermarks against various common signal processing types, while at the same time the proposed scheme has a robust performance.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.160-163
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• 2002

This paper presents a statistical technique for audio watermarking. We describe the application of the promising public key watermarking method to the patchwork algorithm. Its detection process does not need the original content nor the secret key used in the embedding process. Special attention is given to statistical method working in the frequency domain. We will present a solution of robust watermarking of audio data. In this scheme, an extension of patchwork audio watermarking is presented which enables public detection of the watermark. Experimental results show good robustness of the approach against MP3 compression and other common signal processing manipulations.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.4 s.304
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• pp.167-176
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• 2005

This paper proposes a new power minimization technique for the frame-based multimedia signal processing. The derivation of the technique is based on the newly proposed microscopic DVS(Dynamic Voltage Scaling) method, where, the operating frequency and the supply voltage levels are dynamically controlled according to the processing requirement for each frame of multimedia data. The multimedia signal processing algorithms are also redesigned and optimized to maximize the power saving efficiency of the microscopic DVS technology. The characterization of the mean/variance distribution of the processing load in the frame-based multimedia signal processing provides the major basis not only for the optimized application of the microscopic DVS technology but also for the optimization of the multimedia algorithms. The power saying efficiency of the proposed DVS approach is experimentally tested with the algorithms of MPEG-2 video decoder and MPEG-2 AAC audio encoder on the ARM9 RISC processor. The experimental results with the diverse MPEG-2 video and audio files show The average power saving efficiencies of 50$\%$ and 30$\%$, respectively. The results also agree very well with those of the analytic derivations.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.3
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• pp.177-183
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• 2018

Currently, digital signal processing systems are used extensively in image processing, audio processing, filtering, and equalizations, etc. In addition, the importance of DRAM, which has a great influence on the performance of an digital signal processor has been increased, making research on DRAM actively conducted in industry and academia. Therefore, it is important to have a more accurate DRAM model in order to obtain reliable results when evaluating the performance of a digital signal processor through simulation. In this paper, we developed a digital signal processor simulator capable of inter-working with a DRAM simulator. With the simulator, we analyzed the influence of the DRAM model which operates correctly on a cycle-by-cycle basis, on the performance of the digital signal processor by using the UTDSP digital signal benchmark.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.2
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• pp.7-11
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• 2003

Nowadays, as digital technologies are rapidly developed and requirements for the various types of broadband multimedia services increases, the radio broadcasting is moving to digitalization. DAB(Digital Audio Broadcasting), as an alternation of existing analog radio broadcasting, is a new type of multimedia broadcasting system. DAB supports not only high-quality audio broadcasting but also various types of multimedia data services. In this paper, we investigate the performance optimization method of MOT Protocol, as the standard for additional services, to support the multimedia services in the DAB network. Because the performance of the MOT protocol is dependent on various parameters such as segment size, segment repetition and so on, we find those by simulation for performance optimization. According to simulation results, the suitable segment size is about 2Kbyte and segment repetition is 4 times for performance optimization.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.367-374
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• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.2
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• pp.63-68
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• 2015

This paper proposes a robust multiple audio watermarking algorithm using improved QIM(quantization index modulation) with adaptive stepsize for different signal power and SVM(support vector machine) decoding model. The proposed algorithm embeds watermarks into both frequency magnitude response and frequency phase response using QIM. This multiple embedding method can achieve a complementary robustness. The SVM decoding model can improve detection rate when it is not sure whether the extracted data are the watermarks or not. To evaluate robustness, 11 attacks are employed. Consequently, the proposed algorithm outperforms previous multiple watermarking algorithm, which is identical to the proposed one but without SVM decoding model, in PSNR and BER. It is noticeable that the proposed algorithm achieves improvements of maximum PSNR 7dB and BER 10%.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.2
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• pp.233-238
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• 2015

In this paper, we proposed that immersive audio effect method using the effective noise to improve engagement in free-viewpoint TV(FTV) service. In the basketball court, we monitored the frequency spectrums by acquiring continuous audio data of players and referee using shotgun and wireless microphone. By analyzing this spectrum, in case that users zoomed in, we determined whether it is effective frequency or not. Therefore when users using FTV service zoom in toward the object, it is proposed that we need to utilize unnecessary noise instead of removing that. it will be able to be useful for an immersive audio implementation of FTV.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.7
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• pp.308-316
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• 2002

In this paper, we propose a robust digital copyright-protection technique based on the concept of human auditory system. First, we propose a watermarking technique that accepts the various attacks such as, time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC WMA. Second, we implement audio PD(portable device) for copyright protection using proposed method. The proposed watermarking technique is developed using digital filtering technique. Being designed according to critical band of HAS(human auditory system), the digital filers embed watermark without nearly affecting audio quality. Before processing of digital filtering, wavelet transform decomposes the input audio signal into several signals that are composed of specific frequencies. Then, we embed watermark in the decomposed signal (0kHz~11kHz) by designed band-stop digital filer. Watermarking detection algorithm is implemented on audio PD(portable device). Proposed watermarking technology embeds 2bits information per 15 seconds. If PD detects watermark '11', which means illegal song. PD displays "Illegal Song" message on LCD, skips the song and plays the next song, The implemented detection algorithm in PD requires 19 MHz computational power, 7.9kBytes ROM and 10kBytes RAM. The suggested technique satisfies SDMI(secure digital music initiative) requirements of platform3 based on ARM9E core.