• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1C
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• pp.12-18
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• 2008

This paper This paper proposes a new inverse-quantization(IQ) table interpolation algorithm, specialized Digital Signal Processor(DSP) instructions and hardware architecture for digital audio codecs. Non-linear inverse quantization algorithm is representatively used in both MPEG-1 Layer-3 and MPEG-2/4 Advanced Audio Coding(AAC). The proposed instructions are optimized for the non-linear inverse quantization. The proposed algorithm can minimize operational complexity which reduces total computational load. Performance comparisons show a significant improvement of average error. The proposed instructions and hardware architecture can reduce 20% of the instruction counts and minimize computational loads of IQ algorithms effectively compared with existing IQ table interpolation algorithms. Proposed algorithm can implement commercial DSPs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.12C
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• pp.1251-1264
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• 2002

Currently, audio streaming services such as on-demand service and live service support only a limited number of clients. They also suffer from a lack of stability and degradation of service quality due to their inefficient use of network resources. Futhermore, since the streaming services usually do not consider dynamic services, they are very inconvenience to use. In order to resolve these drawbacks, we propose a novel audio streaming system based on MPEG-2 AAC file format which are facilitated with the network bandwidths efficiently. The proposed system supports QoS for audio streaming as well as guarantees a stability while streaming service is undergoing. Moreover, the system provides a dynamic interface which enables us to use the streaming service more easily and to manage streaming servers with convenient manner. On the contrary, most of the current available static interface streaming services are mainly depending only on a single fixed web page between client and server, which in consequence lead us to use unflexible static service environment. Our implementation results show the proposed system improves the performance compared to those of the currently existing systems that use MP3 file format. It also provides some benefits such as a stability of service and a easy to management of streaming servers.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.3 s.309
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• pp.76-83
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• 2006

A main objective of audio equalizer is for user to tailor acoustic frequency response to increase sound comfort and example applications of audio equalizer includes large-scale audio system to portable audio such as mobile MP3 player. Up to now, all the audio equalizer requires manual setting to equalize frequency bands to create suitable sound quality for each genre of music. In this paper, we propose an intelligent audio graphic equalizer system that automatically classifies the music genre using music content analysis and then the music sound is boosted with the given frequency gains according to the classified musical genre when playback. In order to reproduce comfort sound, the musical genre is determined based on two-step hierarchical algorithm - coarse-level and fine-level classification. It can prevent annoying sound reproduction due to the sudden change of the equalizer gains at the beginning of the music playback. Each stage of the music classification experiments shows at least 80% of success with complete genre classification and equalizer operation within 2 sec. Simple S/W graphical user interface of 3-band automatic equalizer is implemented using visual C on personal computer.

• Journal of Intelligence and Information Systems
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• v.17 no.2
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• pp.97-107
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• 2011

Nowadays there are a lot of issues about copyright infringement in the Internet world because the digital content on the network can be copied and delivered easily. Indeed the copied version has same quality with the original one. So, copyright owners and content provider want a powerful solution to protect their content. The popular one of the solutions was DRM (digital rights management) that is based on encryption technology and rights control. However, DRM-free service was launched after Steve Jobs who is CEO of Apple proposed a new music service paradigm without DRM, and the DRM is disappeared at the online music market. Even though the online music service decided to not equip the DRM solution, copyright owners and content providers are still searching a solution to protect their content. A solution to replace the DRM technology is digital audio watermarking technology which can embed copyright information into the music. In this paper, the author proposed a new audio watermarking algorithm with two approaches. First, the watermark information is generated by two dimensional barcode which has error correction code. So, the information can be recovered by itself if the errors fall into the range of the error tolerance. The other one is to use chirp sequence of CDMA (code division multiple access). These make the algorithm robust to the several malicious attacks. There are many 2D barcodes. Especially, QR code which is one of the matrix barcodes can express the information and the expression is freer than that of the other matrix barcodes. QR code has the square patterns with double at the three corners and these indicate the boundary of the symbol. This feature of the QR code is proper to express the watermark information. That is, because the QR code is 2D barcodes, nonlinear code and matrix code, it can be modulated to the spread spectrum and can be used for the watermarking algorithm. The proposed algorithm assigns the different spread spectrum sequences to the individual users respectively. In the case that the assigned code sequences are orthogonal, we can identify the watermark information of the individual user from an audio content. The algorithm used the Walsh code as an orthogonal code. The watermark information is rearranged to the 1D sequence from 2D barcode and modulated by the Walsh code. The modulated watermark information is embedded into the DCT (discrete cosine transform) domain of the original audio content. For the performance evaluation, I used 3 audio samples, "Amazing Grace", "Oh! Carol" and "Take me home country roads", The attacks for the robustness test were MP3 compression, echo attack, and sub woofer boost. The MP3 compression was performed by a tool of Cool Edit Pro 2.0. The specification of MP3 was CBR(Constant Bit Rate) 128kbps, 44,100Hz, and stereo. The echo attack had the echo with initial volume 70%, decay 75%, and delay 100msec. The sub woofer boost attack was a modification attack of low frequency part in the Fourier coefficients. The test results showed the proposed algorithm is robust to the attacks. In the MP3 attack, the strength of the watermark information is not affected, and then the watermark can be detected from all of the sample audios. In the sub woofer boost attack, the watermark was detected when the strength is 0.3. Also, in the case of echo attack, the watermark can be identified if the strength is greater and equal than 0.5.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.261-269
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• 2004

In this paper, an efficient algorithm for computing in the MPEG/audio Layer Ⅲ (MP3) encoder is proposed. The proposed algerian performs a full-band 1024-point FFT by computing 32-point FFT's of 32 subband outputs. To reduce the aliasing caused by the analysis filter bank, an aliasing cancellation butterfly is developed. A major benefit of the proposed algorithm is the computational saving. By using the proposed algorithm, it is possible to save 40~50% of computations for FFT, which results in about 20% reduction of the PAM-2 complexity.

• The Journal of the Korea Contents Association
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• v.2 no.3
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• pp.105-112
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• 2002

MPEG-4 is an international standard for interactive multimedia data that is used for next generation audio-visual service. It presents an two types of storage formal such as XMT (eXtensible MPEG-4 Textual fount) and MP4 file format D store the interactive multimedia contents. The XMT, a textual format, has advantage of readability and it can be used in many applications since designed based on the XML In this paper, we present the way how to author the MPEG-4 contents using XMT.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.569-572
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• 2005

This paper proposed the design and implementation of a DSP platform for the various multimedia streaming. The DSP platform synchronizes with host PC to configure DSP and to transmit multimedia streaming through PCI. The suggested DSP platform decodes high-capacity video/audio data using the suggested high-speed FIFO, CPLD and memory interface. The buffer control techniques is proposed in other to avoid the under/over-run of the audio/video data during the audio/video decoding. For the DSP platform test, host PC transmits program stream(PS) that consists of the MPEG-2 video MP@ML and 5.1ch AC3 audio data (Coyote.mov file, half hour running time) to DSP platform. The DSP platform plays continuously back the high sound-quality audio and high-definition video at once.

• Journal of the Korea Society of Computer and Information
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• v.15 no.2
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• pp.109-117
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• 2010

Digital watermarking has drawn extensive attention for protecting digital contents from unauthorized copying. This paper proposes a new watermarking scheme in frequency domain for copyright protection of digital audio. In our proposed watermarking system, the original audio is segmented into non-overlapping frames. Watermarks are then embedded into the selected prominent peaks in the magnitude spectrum of each frame. Watermarks are extracted by performing the inverse operation of watermark embedding process. Simulation results indicate that the proposed scheme is robust against various kinds of attacks such as noise addition, cropping, resampling, re-quantization, MP3 compression, and low pass filtering. Our proposed watermarking system outperforms Cox's method in terms of imperceptibility, while keeping comparable robustness with the Cox's method. Our proposed system achieves SNR (signal-to-noise ratio) values ranging from 20 dB to 28 dB. This is in contrast to Cox's method which achieves SNR values ranging from only 14 dB to 23 dB.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.653-660
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• 2009

This paper proposes a robust audio watermarking algorithm for copyright protection and improvement of sound quality after embedding a watermark into an original sound. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and divides the spectrum into a subbands. Then, it is necessary to calculate the energy of each subband and sort n subbands in descending order corresponding to its power. After calculating the energy we choose k subbands in sorted order and find p peaks in each selected subbands, and then embed a length m watermark around the p peaks. When the listeners hear the watermarked sound, they do not recognize any distortions. Furthermore, the proposed method is robust as much as Cox's method to MP3 compression, cropping, FFT echo attacks. In addition to this, the experimental results show that the proposed method is generally 10 dB higher than Cox's method in SNR (signal-to-noise ratio) aspect.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.7
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• pp.1657-1670
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• 2013

The increase in the number of music files in smart phone and MP3 player makes it difficult to find the music files which people want. So, Query-by-Singing/Humming (QbSH) systems have been developed to retrieve music from a user's humming or singing without having to know detailed information about the title or singer of song. Most previous researches on QbSH have been conducted using musical instrument digital interface (MIDI) files as reference songs. However, the production of MIDI files is a time-consuming process. In addition, more and more music files are newly published with the development of music market. Consequently, the method of using the more common MPEG-1 audio layer 3 (MP3) files for reference songs is considered as an alternative. However, there is little previous research on QbSH with MP3 files because an MP3 file has a different waveform due to background music and multiple (polyphonic) melodies compared to the humming/singing query. To overcome these problems, we propose a new QbSH method using MP3 files on mobile device. This research is novel in four ways. First, this is the first research on QbSH using MP3 files as reference songs. Second, the start and end positions on the MP3 file to be matched are estimated by using multi-layered perceptron (MLP) prior to performing the matching with humming/singing query file. Third, for more accurate results, four MLPs are used, which produce the start and end positions for dynamic time warping (DTW) matching algorithm, and those for chroma-based DTW algorithm, respectively. Fourth, two matching scores by the DTW and chroma-based DTW algorithms are combined by using PRODUCT rule, through which a higher matching accuracy is obtained. Experimental results with AFA MP3 database show that the accuracy (Top 1 accuracy of 98%, with an MRR of 0.989) of the proposed method is much higher than that of other methods. We also showed the effectiveness of the proposed system on consumer mobile device.