• Journal of Aerospace System Engineering
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• v.11 no.6
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• pp.1-9
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• 2017

The ICS, Intercom is the equipment which mixes and distributes the audio signal from other LRUs and plays the Voice Messages. Henceforth, it is of immense contributory importance to the pilots. Especially, the audio routing, which controls On/Off mode of each audio channel, is significant in executing a pilots' mission. But the audio routing process is quite complicated as it has the interface combination of many control signals. Underthecondition, the exceptional handling becomes difficult, which decreases maintainability and productivity. In the present work, to prevent such a situation, the author suggests amethodology,whichwillhavealower impact when the software is changed and provides high maintainability and productivity for audio routing processing.

• 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.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.12
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• pp.716-722
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• 2005

In this paper, we propose the audio watermarking algorithm based on the critical band of HAS(human auditory system) without audibly affecting the quality of the watermarked audio and implement the detecting algorithm on the BSS(broadcast synchronizing system) for testing the proposed algorithm. According to the audio quality test, the SNR(signal to noise ratio) of the watermarked audio objectively is 66dB above. In the robustness test, the proposed algorithm can detect the watermark more than $90\%$ from various compression(MP3, AAC), A/D and D/A conversions, sampling rate conversions and especially asynchronizing attacks. The BSS automatically switches the programs between the key station and the local station in broadcasting system. The result of reliability test of implemented system by using the real broadcasting audio has no false positive error during 30 days. Because of detecting once processing per 0.5 second, we can judge that the false positive error does not occur.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2101-2102
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• 2006

In this paper, a new digital audio watermarking algorithm is presented. The proposed algorithm embeds watermark into audio signal based on human auditory system (HAS). This algorithm is a blind audio watermarking method, which does not require any prior information during watermark extraction process. This algorithm finds watermarking position using time-domain masking effect. First we insert the watermark into wavelet domain, and then we use a back-propagation neural network (BPN) to learn the characteristics of relationship between the watermark and the watermarked audio. Due to the teaming and adaptive capabilities of the BPN, the false recovery of the watermark can be greatly reduced by the trained BPN. Experimental results show that the proposed method has good inaudibility and high robustness to common audio processing attacks.

• Broadcasting and Media Magazine
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• v.13 no.1
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• pp.82-96
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• 2008

3D Audio Technologies include whole signal processing steps from acquisition to reproduction through encoding and transmitting technologies. However, there is a certain difference on adapted technologies according to audio presentation environments, because the presentation environment is the last step to provide 3D audio th listeners. In this paper, we describe variable 3D audio technologies to adapt variable audio presentation environments for consuming music contents.

• Journal of Broadcast Engineering
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• v.22 no.5
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• pp.632-642
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• 2017

In this paper, we propose a DNN based speech detection system using acoustic characteristics and context information of media audio. The speech detection for discriminating between speech and non-speech included in the media audio is a necessary preprocessing technique for effective speech processing. However, since the media audio signal includes various types of sound sources, it has been difficult to achieve high performance with the conventional signal processing techniques. The proposed method improves the speech detection performance by separating the harmonic and percussive components of the media audio and constructing the DNN input vector reflecting the acoustic characteristics and context information of the media audio. In order to verify the performance of the proposed system, a data set for speech detection was made using more than 20 hours of drama, and an 8-hour Hollywood movie data set, which was publicly available, was further acquired and used for experiments. In the experiment, it is shown that the proposed system provides better performance than the conventional method through the cross validation for two data sets.

• Journal of KIISE:Software and Applications
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• v.34 no.12
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• pp.1037-1044
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• 2007

Recently, we extract the representative melody of the music and index the music to reduce searching time at the content-based music retrieval system. The existing study has used MIDI data to extract a representative melody but it has a weak point that can use only MIDI data. Therefore, this paper proposes a representative melody retrieval method that can be use at all audio file format and uses digital signal processing. First, we use Fast Fourier Transform (FFT) and find the tempo and node for the representative melody retrieval. And we measure the frequency of high value that appears from PCM Data of each node. The point which the high value is gathering most is the starting point of a representative melody and an eight node from the starting point is a representative melody section of the audio data. To verity the performance of the method, we chose a thousand of the song and did the experiment to extract a representative melody from the song. In result, the accuracy of the extractive representative melody was 79.5% among the 737 songs which was found tempo.

• Journal of Broadcast Engineering
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• v.6 no.2
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• pp.195-202
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• 2001

As a solution for copyright protection of digital multimedia contents, digital watermark technology is now drawing the attention. In this paper, we presented two novel audio watermarking algorithms as a solution for protecting unauthorized copy of digital audio. Proposed watermarking schemes include the psychoacoustic model of MPEG audio coding to achieve the perceptual transparency after watermark embedding and preprocessing procedure before correlation in watermark detection to extract copyright information without access to the original audio signal. Experimental results show that our watermarking scheme is robust to common signal Processing attacks and it Introduces no audible distortion after watermark insertion.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.43-48
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• 2000

This paper introduces MPEG-2 AAC codec system fur digital audio broadcasting. This system consists of encoder and decoder, and this system provides MPEG-2 system multiplexing and demultiplexing functions. Four DSPs are adopted fur encoder and three DSPs fur decoder. Each DSP processes system control, I/O control, and audio signal processing, multiplexing and demultiplexing. This paper also discusses about some near future estimations related to DAB system and services. And at the end of this paper describes about future development plans.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1296-1299
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• 2002

We implemented a CD signal processor operated on a CAV 48-speed CD-ROM drive into a VLSI. The CD signal processor is a mixed mode monolithic IC including servo-processor, data recovery, data-processor, and I-bit DAC. For servo signal processing, we included a DSP core, while, for CAV mode playback, we adopted a PLL with a wide recovery range. Data processor (DP) was designed to meet the yellow book specification.［2］So, the DP block consists of EFM demodulator, C1/C2 ECC block, audio processor and a block transferring data to an ATAPI chip. A modified Euclid's algorithm was used as a key equation solver for the ECC block To achieve the high-speed decoding, the RS decoder is operated by a pipelined method. Audio playability is increased by playing a CD-DA disc at the speed of 12X or 16X. For this, subcode sync and data are processed in the same way as main data processing. The overall performance of IC is verified by measuring a transfer rate from the innermost area of disc to the outermost area. At 48-speed, the operating frequency is 210 ㎒, and this chip is fabricated by 0.35 um STD90 cell library of Samsung Electronics.