• 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 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.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.1
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• pp.67-74
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• 2016

In this paper, we propose a watermarking technology for a mixed music. The mixed music means recreated music that contained a number of musics in one audio clip. Royalty associated with the audio content is typically imposed by the full audio content. However, the calculation of royalties gives rise to conflict between copyright holders and users in the mixed music because it uses not full audio content but a fraction of that. To solve the conflict related with the mixed music, we propose a audio watermarking technique that inserts different watermarks for each audio in the audio that make up the mixed music. The proposed watermarking scheme might have poor SNR (signal to noise ratio) to embed to each audio clip. To overcome poor SNR problem, we used inaudible pseudo random sequence which modifies typical pseudo random sequence to canonical signed digit (CSD) form. The proposed method verifies the performance by each watermark extraction and the time internal estimation valies from the mixed music.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.220-226
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• 2000

A novel audio watermark algorithm, the Modified Patchwork Algorithm, is applied to the speech to show that it is effective for digital broadcasting systems. Digital broadcasting system does not separate speech from audio data. However. speech data is very important especially for educational broadcasting. Speech can carry more information than video data. Thus, intellectual property management and protection for speech data is urgent. This paper addresses the technical issues, speech watermark algorithm, and its robustness against malicious attacks.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.10
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• pp.1883-1887
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• 2008

Most of digital media contents are composed with video and audio, picture and animation informations. Sometime, there is some deviation of information recognition quality for the video and audio information according to information receiver's characteristics or the understanding. But visual information using the text provide most clear and accurate ways for information recognition to human being. In this paper, we propose a new broadcasting system(BSDMC) to transmit clear and accurate meaning of the digital media contents. We implement general-purpose components to display the video, picture, text and symbol simultaneously. Only plug-in and call these components with proper parameters on the application developing tool, we can easily develop the multimedia contents broadcasting system. These components are implemented based on the object-oriented framework and modular structure so that increase the reusability and can be develop other applications quick and reliable.

• ETRI Journal
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• v.29 no.1
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• pp.99-102
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• 2007

The channel level difference (CLD) is a main parameter in the reference model 0 (RM0) for MPEG Surround. Nevertheless, the CLD quantization method in the RM0 has problems such as the lack of theoretical background and inappropriate quantization levels. In this letter, a new CLD quantization method is proposed based on the virtual source location information which has strength in the quantization process. From experimental results, it is confirmed that the proposed scheme greatly reduces the quantization distortions measured in dB and degrees without any additional complexity.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.3
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• pp.59-64
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• 2023

Fire and disaster broadcasting systems are divided into analog, digital, and network-based digital public address systems, and important specifications in network-based digital public address systems are low-latency audio, high sampling rate, and multi-channel input and output. In the past, it has been widely used to the AoE method for distinguishing based on the MAC address of the data link layer. However, this method has a problem of increasing complexity and cost. This proposal is an AoIP/UDP method, which allows communication to be easily distinguished by IP address without the need for a separate redundant network, so that the network can be freely used and configured, and cost can be reduced by reducing complexity. After implementing the AoIP/UDP method, the experimental results showed that the cost was improved with the equivalent performance with 2.66ms latency.

• 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.

• Journal of IKEEE
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• v.3 no.1 s.4
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• pp.11-21
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• 1999

This paper has been proposed a structure composed of FIRs and IIR filters as digital interpolation filter to integrate the off-chip analog low-pass filter of audio DAC. The passband ripple (>$0.41{\times}fs$), passband attenuation(>at$0.41{\times}fs$) and stopband attenuation(<$0.59{\times}fs$) of the ${\Delta}{\Sigma}$ modulator output using the proposed digital interpolation filter had ${\pm}0.001[dB]$, -0.0025[dB] and -75[dB], respectively. Also the inband group delay was 30.07/fs[s] and the error of group delay was 0.1672%. Also, the attenuation of stopband has been increased -20[dB] approximately at 65[kHz], out-of-band. Therefore the RC products of analog low-pass filter on chip have been decreased compared with the conventional digital interpolation filter structure.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.4
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• pp.156-163
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• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.