• Journal of Digital Contents Society
• /
• v.15 no.3
• /
• pp.357-363
• /
• 2014

In this paper, we propose a new audio forensic marking system for protecting the copyright of the Internet-based music streaming services. In the proposed method, in order to guarantee the QoS of the streaming service, high speed, and generates a forensic mark inserted MP3 file. We make pre-marking process and generate a new forensic marked MP3 file, a combination of the pre-marked MP3 frame, the inserted user information. Experimental results show that the proposed method satisfactory results robustness and imperceptibility, and real-time properties. In addition, we were confirmed that the real-time embedding and detection from the streaming-based audio forensic marking system that has been implemented on the server/client is possible.

• Journal of Korea Society of Industrial Information Systems
• /
• v.11 no.5
• /
• pp.18-27
• /
• 2006

Rapid increase of the amount of music data demands for a new method that allows efficient similarity retrieval of music genre using audio features in music databases. To build this similarity retrieval, an indexing techniques that support audio features as a time-series pattern and data mining technologies are needed. In this paper, we address the development of a system that retrieves similar genre music based on the indexing techniques. We first propose the structure of content-based music genre retrieval system based on the time-series pattern index file and data mining technologies. In addition, we implement the time-series pattern index file using audio features and present performance analysis of the time-series pattern index file for similar genre retrieval. The experiments are performed on real data to verify the performance of the proposed method.

• IEMEK Journal of Embedded Systems and Applications
• /
• v.2 no.3
• /
• pp.184-193
• /
• 2007

Flash memory adoption in the mobile devices is increasing or vanous multimedia services such as audio, videos, and games. Although the traditional research issues such as out-place update, garbage collection, and wear-leveling are important, the performance, memory usage, and fast mount issues of flash memory file system are becoming much more important than ever because flash memory capacity is rapidly increasing. In this paper, we address the problems of the existing log-based flash memory file systems analytically and propose an efficient log-based file system, which produces higher performance, less memory usage and mount time than the existing log-based file systems. Our ideas are applied to a well-known log-based flash memory file system (YAFFS2) and the performance tests are conducted by comparing our prototype with YAFFS2. The experimental results show that our prototype achieves higher performance, less system memory usage, and faster mounting than YAFFS2, which is better than JFFS2.

• Proceedings of the IEEK Conference
• /
• 2004.08c
• /
• pp.768-771
• /
• 2004

In this paper, we propose a content-based audio genre classification algorithm that automatically classifies the query audio into five genres such as Classic, Hiphop, Jazz, Rock, Speech using digital signal processing approach. From the 20 second query audio file, 54 dimensional feature vectors, including Spectral Centroid, Rolloff, Flux, LPC, MFCC, is extracted from each query audio. For the classification algorithm, k-NN, Gaussian, GMM classifier is used. In order to choose optimum features from the 54 dimension feature vectors, SFS (Sequential Forward Selection) method is applied to draw 10 dimension optimum features and these are used for the genre classification algorithm. From the experimental result, we verify the superior performance of the SFS method that provides near $90{\%}$ success rate for the genre classification which means $10{\%}$-$20{\%}$ improvements over the previous methods

• Phonetics and Speech Sciences
• /
• v.6 no.2
• /
• pp.21-28
• /
• 2014

We propose a novel scheme for digital audio authentication of given audio files which are edited by inserting small audio segments from different environmental sources. The purpose of this research is to detect inserted sections from given audio files. We expect that the proposed method will assist human investigators by notifying suspected audio section which considered to be recorded or transmitted on different environments. GMM-UBM and GSV-SVM are applied for modeling the dominant environment of a given audio file. Four kinds of likelihood ratio based scores and SVM score are used to measure the likelihood for a dominant environment model. We also use an ensemble score which is a combination of the aforementioned five kinds of scores. In the experimental results, the proposed method shows the lowest average equal error rate when we use the ensemble score. Even when dominant environments were unknown, the proposed method gives a similar accuracy.

• Proceedings of the IEEK Conference
• /
• 1998.10a
• /
• pp.1215-1218
• /
• 1998

본 논문은 Digital Audio Compression(AC-3) Standard 인 A-52를 기반으로 하였으며 Borland C++3.1 Compiler를 사용하여 AC-3 Decoding Algorithm 구현하였다. Input Stream은 DVD VOB File에서 AC-3 Stream만을 분리하여 사용하며 최종 출력은 16 Bit PCM File이다. AC-3의 Frame구조는 Synchronization Information, Bit Stream Information, Audio Block, Auxiliary Data, Error Check로 구성된다. Aduio Block 은 모두 6개의 Block으로 나뉘어져 있다. BSI와 Side Information을 참조하여 Exponent를 추출하여 Exponent Strategy에 따라 Exponent를 복원한다. 복원된 Exponent 정보를 이용하여 Bit Allocation을 수행하여 각각의 Mantissa에 할당된 Bit수를 계산하고 Stream으로부터 Mantissa를 추출한다. Coupling Parameter를 참조하ㅕ Coupling Channel을 Original Channel로 복원시킨다. Stereo Mode에 대해서는 Rematrixing을 수행한다. Dynamic Range는 Mantissa와 Exponent의 Magnitude를 바꾸는 것으로 선택적으로 사용할 수 있다. Mantissa와 Exponent를 결합하여 Floating Point coefficient로 만든 후 Inverse Transform을 수행하면 PCM Data를 얻을 수 있다. PC에서 듣기 위해서는 Multi Channel을 Stereo나 Mono로 Downmix를 수행한다. 이렇게 만들어진 PCM data는 PCM Data를 재생하는 프로그램으로 재생할 수 있다.

• Proceedings of the KIEE Conference
• /
• 2004.07d
• /
• pp.2429-2431
• /
• 2004

In this paper, we have designed a remote controlable HDR System using all embedded linux board. The system is composed of three parts - a HDR System, a PC client program for remote control and a Nameserver for registering and aquisition of the IP address. The system is built in an embedded board using a linux kernel. With the Linux the system can support networking and file system for a hard disk management In addition, the system embeds a web-server and a ftp-server for remote manipulation and file transfer. And the hardwares of the system are controlled by the linux device driver mechanism. MPEG1/2 technique is used to compress TV tuner signal and external analog video/audio signal. And compressed data is stored in a hard disk. The data stored in the system is accesable through lan or internet. And RTP protocol is used to enable the system to service live stream of instant video/audio input.

• The KIPS Transactions:PartB
• /
• v.9B no.1
• /
• pp.67-76
• /
• 2002

MP3 audio format has good quality and high compression rate ; therefore, the use of MP3 format increases. The requirement of keeping the extra information such as author and lyrics in MP3 files increases. And the tagging systems designed to meet this requirement are suggested. ID3 vl tag and Lyrics3 v2 tag are two most widely used tagging systems. But ID3 vl tag and Lyrics3 v2 tag are the last things to arrive when the file is being streamed. Therefore, users cannot get the tag information until the entire audio file is downloaded. Moreover information synchronized with audio stream may lose its feature. In this paper, a system searching and playing audio files based on tag information in MP3 files is implemented. This system solves the problem that the tag information is ignored when an MP3 files is played on internet. An audio object is described in an XML document, and timing and synchronization between elements in that In document is provided in HTML+TIME style using XSL.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.2
• /
• pp.136-143
• /
• 2017

In this paper, we propose a method for restoring damaged audio files using deep neural network. It is different from the conventional file carving based restoration. The purpose of our method is to infer lost information which can not be restored by existing techniques such as the file carving. We have devised methods that can automate the tasks which are essential for the restoring but are inappropriate for humans. As a result of this study it has been shown that it is possible to restore the damaged files, which the conventional file carving method could not, by using tasks such as speech or nonspeech decision and speech encoder recognizer using a deep neural network.

• Journal of the Korea Institute of Information Security & Cryptology
• /
• v.27 no.3
• /
• pp.541-548
• /
• 2017

The smartphone operates the media server daemon to handle audio service requests. Media server daemons, running with a high privilege in the background, caused many vulnerabilities to applications most frequently used in smart devices including smartphones. Fuzzing is a popularly used methodology to find software vulnerabilities. Unfortunately, fuzzing itself is not much effective in such format-strict environments as media services. In this paper, we propose a file format-aware fuzzing in order to efficiently detect vulnerabilities of media server daemon. We acquired a remote arbitrary code execution vulnerability on iOS/tvOS/MacOS/watchOS, and we verified the effectiveness by comparing our methodology with the fuzzers FileFuzz and ZZUF.