• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.275-278
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• 2004

In TV sound system, loudspeakers are subject to severe size constraints. The small size of the transducer affects the low frequency signal performance of the system. Bass signal performance contributes significantly to the user perceived sound quality and a good bass signal reproduction is essential. Increasing the sound energy in the bass signal range is an unviable solution since the gain required are exceedingly high and signal distortion occurs because of the speaker overload. Recently methods are being proposed to invoke low frequency illusion using psychoacoustic phenomena of the missing fundamental. This paper proposes a simple and effective signal processing method to create bass signal illusion in TV speakers using the missing fundamental effect, at a complexity of 12 MIPS on Motorola 56371 audio DSP.

• Journal of Information Processing Systems
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• v.12 no.3
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• pp.525-537
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• 2016

Data hiding is a wide field that is helpful to secure network communications. It is common that many data hiding researchers consider improving and increasing many aspects such as capacity, stego file quality, or robustness. In this paper, we use an audio file as a cover and propose a reversible steganographic method that is modifying the sample values using modulus function in order to make the reminder of that particular value to be same as the secret bit that is needed to be embedded. In addition, we use a location map that locates these modified sample values. This is because in reversible data hiding it needs to exactly recover both the secret message and the original audio file from that stego file. The experimental results show that, this method (measured by correlation algorithm) is able to retrieve exactly the same secret message and audio file. Moreover, it has made a significant improvement in terms of the following: the capacity since each sample value is carrying a secret bit. The quality measured by peak signal-to-noise ratio (PSNR), signal-to-noise ratio (SNR), Pearson correlation coefficient (PCC), and Similarity Index Modulation (SIM). All of them have proven that the quality of the stego audio is relatively high.

• Journal of Advanced Navigation Technology
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• v.10 no.4
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• pp.406-412
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• 2006

In this paper, we analyzed the performance of digital watermarking by using audio signal as well as logo image watermark. By utilizing the OFDM/QPSK system under AWGN channel environment, watermarked image are transmitted and detected. Experimental results showed that audio signal-based watermark embedding scheme is superior to that of logo image-based, which is able to restore a signal at SNR=3[dB].

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.1133-1136
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• 2005

The MPEG/audio standard results from three years of co-work by an international committe of high-fidelity audio compression experts in the Moving Picture Experts Group (MPEG/audio). The MPEG standard is rigid only where necessary to ensure interoperability. In this paper, a new approach of training and classification of acoustic signal is addressed. This is some what a fields of application aspects rather than technonical problems such as MPEG/codec, MIDI. In preprocessing, acoustic signal is transformmed using DWT so as to extract a feature parameters of sound such as loudness, pitch, bandwidth and harmonicity. these accoustic parameters are exploited to the input vector of neural perceptron. Experimental results showed that proposed approach can be used for tunning the dissonance chord.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.155-162
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• 2004

This paper concentrated on lip synchronization (lip sync) test for DTV with respect to audio and video signals using a finite digital bitstream In this paper, we propose a new lip sync test method which does not effect on the current program by use of the transient effect area test signals (TATS) and audio-video time index lip sync test signals (TILS).the experimental result shows that the time difference between audio and video signal can be easily measured by captured oscilloscope waveform at any time.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.24 no.12
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• pp.147-155
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• 2010

We studied on performance improvement of TTM(TI21 Test Meter) that is test and diagnosis devices for jointless audio frequency track circuit on Korean electric railway TI21 standard. Upgraded devices is AD-TTM(Advanced TI21 Test Meter). This can measure alternating frequency USB(Upper signal band) and LSB(Lower signal band). In the audio frequency track circuit, ${\pm}17[Hz]$ of nominal frequency are demodulated and supplied to track relay through AND gate. It is important that measurement function which is error between USB and LSB. Need of AD-TTM will stand out in the electric railway system because this is simple and accurate rather than the former device.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.150-152
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• 2004

Conventional CDDA (Compact Disc Digital Audio) system has limitations come from sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. So, new medium is developed for high-resolution audio recording, like as DVD-audio etc. But CDDA is a widely used medium for high fidelity audio yet, because new medium has complexity and difficulty in manufacturing system. In this paper, we design a new encoding system for high-resolution audio signal. The system is backward compatible with conventional CDDA. By evaluation for encoding and decoding process, we describe practicability of our proposal system.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.5
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• pp.327-329
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• 2005

Conventional CDDA(Compact Disc Digital Audio) system has limitation which means that bandwidth and resolution of the sign릴 are determined by the sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. Though, new medium such as DVD-audio is developed for high-resolution audio recording, it has high complexity and difficulty in manufacturing process. So, CDDA is a widely used medium for high fidelity audio yet. In this paper, we design a new encoding system for high-resolution audio signal which has backward compatible with conventional CDDA. By evaluating for the encoding and decoding process. we verify the availability of our proposed system.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.49 no.4
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• pp.16-22
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• 2012

In this paper, the forensic marking algorithm is proposed using psychoacoustic model II and MDCT for high-quality audio. The proposed forensic marking method, that inserts the user fingerprinting code of the audio content into the selected sub-band, in which audio signal energy is lower than the spectrum masking level. In the range of the one frame which has 2,048 samples for FFT of original audio signal, the audio forensic marking is processed in 3 sub-bands. According to the average attack of the fingerprinting codes, one frame's SNR is measured on 100% trace ratio of the collusion codes. When the lower strength 0.1 of the inserted fingerprinting code, SNR is 38.44dB. And in case, the added strength 0.5 of white gaussian noise, SNR is 19.09dB. As a result, it confirms that the proposed audio forensic marking algorithm is maintained the marking robustness of the fingerprinting code and the audio high-quality.