• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.45-55
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• 2010

Object-based audio coding technology has been interested with its expectation to apply in wide areas. Recently, ISO/IEC MPEG has standardized a parametric object audio coding method, the SAOC (Spatial Audio Object Coding). This paper introduces parametric object audio coding techniques with special focus on the MPEG SAOC and also describes several issues and solutions that should be considered for a success in its application.

• ETRI Journal
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• v.38 no.5
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• pp.829-837
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• 2016

In this paper, we introduce extension tools for MPEG Surround, which were recently adopted as MPEG-H 3D Audio tools by the ISO/MPEG standardization group. MPEG-H 3D Audio is a next-generation technology for representing spatial audio in an immersive manner. However, considerably large numbers of input signals can degrade the compression performance during a low bitrate operation. The proposed extension of MPEG Surround was basically designed based on the original MPEG Surround technology, where the limitations of MPEG Surround were revised by adopting a new coding structure. The proposed MPEG-H 3D Audio technologies will play a pivotal role in dramatically improving the sound quality during a lower bitrate operation.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.630-639
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• 2009

This paper introduces MPEG SAOC (Spatial Audio Object Coding) that has been standardized in MPEG audio subgroup. MPEG SAOC is a trendy parametric coding technology conceptually similar to PS (Parametric Stereo) and the MPEG Surround. SAOC especially parameterizes and codes the spatial information for the object signals comprising a downmixed audio scene and thus lets users render one's preferred scene in an interactive manner.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.25-33
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• 2010

Multi-media service has been changed into user based audio services, which service supports actively user's preference and interaction with the users. In the market, multi-media products which can support the highest audio-quality by using lossless audio technology have been released and object audio music which user can select the objects has been serviced. In this paper, we design user's preference information based object audio file format and audio scene description for storage and transmission media. The designed file format is designed based on MPEG-4 file format because high-quality audio codecs in MPEG-4 audio can be easily used and the track of file format can be flexibly controlled depend on the number of the instrument in music. The encoded audio data of each objects and encoded audio scene description by binary encoding that has independent track are packed in a file. The scene description for storage media is consist of full and object scene description, the scene description for transmission media has an essential description for object audio operation and a specific description for real audio sound. The designed file format based simulator is developed and it generates an object audio file with several scene descriptions. Also, the real audio sound is serviced by the interaction with user and the unpacked scene description.

• Broadcasting and Media Magazine
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• v.4 no.1
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• pp.62-79
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• 1999

In this survey paper the emerging MPEG-4 audio technology is discribed In the previous MPEG-1 and the MPEG-4 audio words, only the natural audio and the speech coding techniques were the standadization objects But in the MPEG-4 audio standadization, not only the natural audio and the speech coding, but also the structured audio and the synthetic speech techniques are inclued, The purpose of this expansion can be summarized as the preparation for the versatile high-quality multimedia services supposed emerge in the 21st century.

• Journal of information and communication convergence engineering
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• v.5 no.3
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• pp.245-248
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• 2007

In this paper, we research the content-based analysis and classification according to the composition of the feature parameter data base for the audio data to implement the audio data index and searching system. Audio data is classified to the primitive various auditory types. We described the analysis and feature extraction method for the feature parameters available to the audio data classification. And we compose the feature parameters data base in the index group unit, then compare and analyze the audio data centering the including level around and index criterion into the audio categories. Based on this result, we compose feature vectors of audio data according to the classification categories, and simulate to classify using discrimination function.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.15 no.2
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• pp.522-539
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• 2021

In order to solve the problems of the existing audio fingerprint method when extracting audio fingerprints from long speech segments, such as too large fingerprint dimension, poor robustness, and low retrieval accuracy and efficiency, a robust audio fingerprint retrieval method based on feature dimension reduction and feature combination is proposed. Firstly, the Mel-frequency cepstral coefficient (MFCC) and linear prediction cepstrum coefficient (LPCC) of the original speech are extracted respectively, and the MFCC feature matrix and LPCC feature matrix are combined. Secondly, the feature dimension reduction method based on information entropy is used for column dimension reduction, and the feature matrix after dimension reduction is used for row dimension reduction based on energy feature dimension reduction method. Finally, the audio fingerprint is constructed by using the feature combination matrix after dimension reduction. When speech's user retrieval, the normalized Hamming distance algorithm is used for matching retrieval. Experiment results show that the proposed method has smaller audio fingerprint dimension and better robustness for long speech segments, and has higher retrieval efficiency while maintaining a higher recall rate and precision rate.

• 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.