• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1127-1130
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• 1999

In this paper, using one method of Additive Synthesis, Analysis-by-synthesis/Overlap-Add (ABS/OLA) method, analysis and synthesis of musical tones is processed. But peak detection of frequency domain is processed by proposed method considering the view of acoustics. It is that that harmonics frequency is times of main frequency. Using this fact, peak detection of frequency domain is useful for detection of tonal component identified musical note. It is possible to realize high-quality lour bit rate audio.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.210-216
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• 2013

It is hard to listen to the music clearly in the presence of background noise. In this paper, a method that modifies the audio signal automatically to enhance the audio listening experience in adverse environment is proposed. Specifically, the method that amplifies the audio signal so that the perceived loudness of audio signal in each band becomes similar to that of the noiseless signal. The loudness perception model proposed by Moore et. al is utilized. Extending the previous work that is applied to speech reinforcement, the full band signal sampled at 48kHz is manipulated based on the loudness restoration principle. Moreover, based on the observation that the audio clarity is compromised even with loudness restored signal, a modification that intentionally boosts high frequency loudness more than lower band is also proposed. Experimental results showed that the proposed algorithm can enhance the audio listening experience in adverse environment.

• 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.

• Journal of Advanced Navigation Technology
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• v.26 no.5
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• pp.358-364
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• 2022

The Inter Communication System for avionics is in charge of processing all voice signals that internal calls between Pilot and Co-pilot, internal calls between Pilots and Crews, external calls through communication equipment such as Ultra/Very High Frequency Receiver/Transmitter(U/VHF RT), audio signal monitoring for navigation and mission equipment such as VHF Omnidirectional Range/Instrument Landing System(VOR/ILS), Tactical Air Navigation(TACAN), audio signal output for voice recording to Flight Data Recorder(FDR) and Data Transfer System(DTS), and warning/caution audio signal generate about the status and threat of aircraft. Because Inter Communication System for avionics is sensitive to noise in the case of analog audio signals, a redundant design that can protect audio signal from electromagnetic noise inside/outside of aircraft is required for the mission of pilots and crews. In this paper, Normal/Back-up operation mode and redundancy design plan based on digital method for the redundancy of the digital Inter Communication System for avionics and manufacturing, verification results are described.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.16 no.5
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• pp.239-246
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• 2023

In this paper, we propose an open-loop algorithm to classify speech and music signals using the spectral flux parameters and Mel Frequency Cepstral Coefficients(MFCC) parameters for the audio coder. To increase responsiveness, the MFCC was used as a short-term feature parameter and spectral fluxes were used as a long-term feature parameters to improve accuracy. The overall voice/music signal classification decision is made by combining the short-term classification method and the long-term classification method. The Gaussian Mixed Model (GMM) was used for pattern recognition and the optimal GMM parameters were extracted using the Expectation Maximization (EM) algorithm. The proposed long-term and short-term combined speech/music signal classification method showed an average classification error rate of 1.5% on various audio sound sources, and improved the classification error rate by 0.9% compared to the short-term single classification method and 0.6% compared to the long-term single classification method. The proposed speech/music signal classification method was able to improve the classification error rate performance by 9.1% in percussion music signals with attacks and 5.8% in voice signals compared to the Unified Speech Audio Coding (USAC) audio classification method.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.215-217
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• 2008

The most widely used method for searching contents of mot ion picture compares contents by extracted cuts. The cut extract ion methods, such as CHD(Color Histogram Difference) or ECR(Edge Change Ratio), are very weak at modifications such as cropping, resizing and low bit rate. The suggested method uses audio contents for indexing and searching to make search be robust against these modification. Scenes of audio contents are extracted for modification-robust search. And based on these scenes, make spectral powers binary on each frequency bin. in the time-frequency domain. The suggested method shows failure rate less than 1% on the false positive error and the true negative error to the modified(using cropping, clipping, row bit rate, addtive frame) contents.

• IEIE Transactions on Smart Processing and Computing
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• v.3 no.5
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• pp.259-266
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• 2014

In this paper, we propose a novel technique for noise robust automatic speech recognition (ASR). The development of ASR techniques has made it possible to recognize isolated words with a near perfect word recognition rate. However, in a highly noisy environment, a distinct mismatch between the trained speech and the test data results in a significantly degraded word recognition rate (WRA). Unlike conventional ASR systems employing Mel-frequency cepstral coefficients (MFCCs) and a hidden Markov model (HMM), this study employ histogram of oriented gradient (HOG) features and a Support Vector Machine (SVM) to ASR tasks to overcome this problem. Our proposed ASR system is less vulnerable to external interference noise, and achieves a higher WRA compared to a conventional ASR system equipped with MFCCs and an HMM. The performance of our proposed ASR system was evaluated using a phonetically balanced word (PBW) set mixed with artificially added noise.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.245-248
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• 1998

In this paper, a new variable wavelet packet decomposition audio coder, based on the time varying characteristic of the audio signals, is proposed and presents a technique to incorporate psychoacoustic models into an adaptive wave let packet scheme. The proposed filterbank improves the defect of the polyphase filterbank that could not properly represent the critical band and the defect of QMF-tree filter that need high complexity to implement. The filterbank consists of varying number of subband from 4 to 26 bands and use Daubechies 6-order wave let. The codec yields excellent quality at total bit rates of about 128kbps for monophonic CD-quality signals with an sampling frequency of 44.1kHz and reduces complexity by 19% for various bit-rates and sources with encoding and decoding process.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.39-44
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• 2001

We investigate the numerical instability of linear prediction for discrete harmonic spectra of audio signals. It is identified that the eigenvalue spread is very large when discrete harmonic spectra are confined only in a lower part of the entire signal bandwidth. A simple method that redefines the sampling frequency and associate harmonic frequencies is proposed to improve the numerical stability. Simulation results using real audio signals indicate its superior stabilizing ability and improved accuracy in the discrete spectral estimation for both LP and DAP.

• Transactions of the Korean Society of Automotive Engineers
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• v.4 no.4
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• pp.1-8
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• 1996

The sound quality of the car audio system is affected by the serveral factors such as the dimensions of the room, the boundary condition of the wall, the location of the speakers, etc. Among these factors, the location of the car speakers has been focused to find the best location of the car speakers assuming that the flat response is better. To verify the suggestion, the subjective test is adopted using 10 people. The developed method is utilizd to evaluate the function of the audio system with fixed speaker position.