• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.66-76
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• 2001

Conventional echo watermarking techniques often exhibit inherent trade-offs between imperceptibility and robustness. In this paper, a new echo embedding technique is proposed. The proposed method enables one to embed high energy echoes while the host audio quality is not deteriorated, so that it is robust to common signal processing modifications and resistant to tampering. It is possible due to echo kernels that are designed based on psychoacoustic analyses. In addition, we propose some novel techniques to improve robustness against signal processing attacks. Subjective and objective evaluations confirmed that the proposed method could improve the robustness without perceptible distortion.

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

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.63-67
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• 1997

The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.80-88
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• 1997

In this paper, the implementation of audio/data processor using ETRI DSP to support analog mode in digital/analog dual mode mobile phone is presented. Audio/data processor performs the wideband data processing, audio signal processing, demodulation function, and data rate conversion when it is operated in analog mode. These functions are programmed in assembly language, and then loaded to ETRI DSP together with vocoder program for the digital mode operation. This is a very efficient implementation of the dual mode cellular phone ASIC since the vocoder for the digital mode and audio/data processor for the analog mode are programmed together in the same hardware.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.

• International conference on construction engineering and project management
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• 2022.06a
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• pp.736-743
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• 2022

Construction workers experience a high rate of fatal incidents from mobile equipment in the industry. One of the major causes is the decline in the acoustic condition of workers due to the constant exposure to construction noise. Previous studies have proposed various ways in which audio sensing and machine learning techniques can be used to track equipment's movement on the construction site but not on the audibility of safety signals. This study develops a novel framework to help automate safety surveillance in the construction site. This is done by detecting the audio sound at a different signal-to-noise ratio of -10db, -5db, 0db, 5db, and 10db to notify the worker of imminent dangers of mobile equipment. The scope of this study is focused on developing a signal processing model to help improve the audible sense of mobile equipment for workers. This study includes three-phase: (a) collect audio data of construction equipment, (b) develop a novel audio-based machine learning model for automated detection of collision hazards to be integrated into intelligent hearing protection devices, and (c) conduct field experiments to investigate the system' efficiency and latency. The outcomes showed that the proposed model detects equipment correctly and can timely notify the workers of hazardous situations.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• Proceedings of the KSLP Conference
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• 1995.11a
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• pp.154-164
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• 1995

CSL, Model 4300B is a highly flexible audio processing package designed to provide a wide variety of speech analysis operations for both new and sophisticated users. Operations include 1) Data acquisition 2) File management 3) Graphics 4) Numerical display 5) Audio output 6) Signal editing 7) A variety of analysis functions, External module include 1) Input control B) Output control 3) Jacks, Software include 1) Wide range of speech display manipulation 2) Editing 3) Analysis (omitted)

• Journal of Advanced Marine Engineering and Technology
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• v.38 no.9
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• pp.1125-1130
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• 2014

In this paper, we propose a sound reception system against surrounding noise for ships based on digital signal processing technologies. In order to suppress unwanted surrounding noises, a digital band-pass filter is designed, which the pass-band of the filter is between 70Hz to 820Hz. Also, we develope a sound direction indicating algorithm with 4 microphones. After filtering the audio signals from 4 microphones, the developed sound direction indicating algorithm can indicate 8 directions. In addition, we implement prototype board for the sound reception using a digital signal processor chip and audio codecs, and verify the proposed algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.139-146
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• 2009

Parametric array for audio applications is analyzed by numerical modeling and analytical approximation. The nonlinear wave equations are used to provide design guidelines for the audio parametric array. A time domain finite difference code that accurately solves the KZK (Khokhlov-Zabolotskaya-Kuznetsov) nonlinear parabolic wave equation is used to predict the response of the parametric array. The time domain code relates the source size and the carrier frequency to the audible signal response including the output level and beamwidth to considering the implementation issues for audio applications of the parametric array, the emphasis is given to the frequency response and distortion. We use the time domain code to find out the optimal parameters that will help produce the parametric array with highest achievable output in terms of the average power within the demodulated signal. Parameters such as primary input frequency, audio source radius and the modulation method are given utmost importance. The output effect of those parameters are demonstrated through the numerical simulation.