Earticle

This paper presents a method of fault detection and fault localization from acoustic noise measurements. In order to detect the presence of noise sources wavelet transform is applied to acoustic signal. In addition, a cross correlation based method is proposed to calculate the exact location of the noise allowing the user to quickly diagnose and resolve the source of the noise. The fault detection system is implemented using two microphones and a computer system. Experimental results show that the system can detect faults due to artifacts accidentally inserted during the manufacturing process and estimate the location of the fault with approximately 1 cm precision.

음향 전력 증폭기의 출력단에서 발생하는 비선형성을 보상하는 디지털 신호 처리 기술을 제안하고 그 모의실험 결과를 제시한다. 음향 전력 증폭기에 사용되는 소자에 의한 비선형성을 간접학습구조와 적응형 필터로 구성되는 디지털 신호 처리 기술로 보상한다. 적응형 필터를 사용함으로써 증폭기의 비선형 특성이 시간적으로 변하더라도 이를 적응적으로 보상할 수 있다. 모의실험 결과 전치 보상기는 3 차의 다항식으로 구현할 수 있으며 홀수 차 비선형성을 효과적으로 제거할 수 있음을 보였다. 짝수 차 비선형은 출력 신호에 존재하는 직류 옵셋이 가장 큰 부분을 차지하며 이는 제안하는 기술로는 제거가 어려우므로 바이어스 회로 설계 시 유의해야 한다. 제안하는 기술은 아날로그 시스템의 본질적 특성 결함을 디지털 신호 처리 기술로서 보상할 수 있음을 보여준다.

We propose a digital signal processing technic that can compensate the non-linearity inherent in audio amplifiers, and present the result of the simulation. The inherent non-linearity of the audio power amplifier arising from analog devices is compensated via a digital signal processing technic consisting of indirect learning architecture and an adaptive filter. The simulation results show that the compensator can be realized using a third-order polynomial and compensates odd-order non-linearity efficiently. The even-oder non-linearity is mainly due to the dc offset at the output, which is difficult to eliminate with the proposed method. Care must be taken in designing the bias circuit to avoid the DC offset at the output. The proposed technic has significance in that digital signal processing technic can compensate for the impairment that is an inherent characteristic of an analog system.

A novel device which can play audio files with LED lights whose RGB effects are based on frequency spectrum analysis to the audio signal while it is playing is introduced in this paper. Not only can the device bring us with songs, but also it presents colorful lights while we appreciate music. RGB light mixed principle and the relations between music and light colors which are through LED lights driven by PWM are investigated. The device is composed of SD card, U disk to store medium files, the audio decoder to produce audio signals, speakers etc., and the all components are under the control of a microprocessor. The proper sampling rate is chosen for frequency analysis, and the Fast Fourier Transform (FFT) method is optimized to suite for the application. File system is implanted to the microprocessor system which possesses wireless communication Xbee, whose functions such as medium file selection can be controlled remotely.

Information hiding technique is a new kind of secret communication technology. The majority of today’s information hiding systems uses multimedia objects like audio. Embedding secret messages in digital sound is usually a more difficult process. Varieties of techniques for embedding information in digital audio have been established. In this paper we will attend the general principles of hiding secret information using audio technology, and an overview of functions and techniques.

In this paper we propose the LSB technique for secure data transfer. Secret information can be hidden inside all sorts of cover information: text, images, audio, video and more. Embedding secret messages in digital sound is usually a more difficult process. Varieties of techniques for embedding information in digital audio have been established. These are parity coding, phase coding, spread spectrum, echo hiding, LSB. Least significant bits (LSB) insertion is one of the simplest approaches to embedding information in audio file.

The sound we hear is transmitted through the atmosphere. However, both the sound we want to hear and the surrounding sound are mixed, and noise is generated, and the sound is not clearly transmitted due to factors such as distance. In particular, in closed spaces like buildings, it is often difficult to hear sounds from outside because of the sound of reflection. People with hearing impairments, such as the elderly and the deaf, have a hard time hearing the sounds they want to hear. Thus, we are developing a hearing aid that can detect radio waves. To this end, we propose the development of a hearing aid that uses FM radio and Bluetooth. These devices are expected to be useful not only for the elderly and the deaf but also in situations where information is transmitted to a large number of people, such as students and tourists, in a large space. The main purpose of this device is to enable users to hear sound correctly without blind spots.

최근에 오디오 신호의 디지털 전송기술의 발전 및 디지털 전송기술을 이용한 오디오 네트워크 장비들의 출시 가 이루어지고 있다. 이에 따라 음향시스템의 설계 및 시공에 있어서도 오디오 네트워크 기술 및 장비의 적용이 적극 적으로 이루어지고 있다. 회의실이라는 공간은 다수의 참가자가 상호의견교환 및 의사전달을 하는 공간으로, 발언내용 이 참석자에게 잘 전달되어야 한다. 마이크 및 스피커 등의 전기음향 장치를 이용하는 것 만 아니라 오디오 네트워크 를 이용한 사례를 통해 회의실의 명료도 향상을 개선하고 실제 사례를 통해 오디오 네트워크를 이용한 음향시스템 설 계의 적용과 향후 발전방향을 전망한다.

Recently, the development of digital transmission technology of audio signals and the introduction of audio network equipment using digital transmission technology have been made. As a result, audio network technology and equipment are actively applied to the design and construction of audio systems. The meeting room is a place where a large number of participants exchange opinions and communicate with each other. In addition to using an electric acoustic device such as a microphone and a speaker, it improves the intelligibility of the conference room through an example using an audio network.

본 논문에서는 파라메트릭 어레이를 이용한 음향신호에 대한 수치 모델링 기법 및 분석 결과를 제시한다. 사용된 음성 파라메트릭 배열의 분석 수치모델은 KZK(Khokhlov-Zabolotskaya-Kuznetsov)로서 KZK수치모델은 시간영역의 차분방정식 알고리즘을 사용하며 파라메트릭배열의 정확한 응답특성이 분석이 가능하다. 시간영역기반의 KZK모델은 음원의 크기와 전송주파수의 영향을 받으며, 가청신호응답은 출력레벨과 빔폭의 크기를 포함한다. 음성신호에 대하여 파라메트릭 배열을 효율적으로 적용시키기 위해서는 고려해야할 요소는 표본화 주파수, 트랜스듀서의 반경 및 변조방식 파라미터 등이 있다. 본 논문에서는 다양한 요소 중 표본화 주파수에 따른 응답신호의 왜곡 분석 및 실험 결과를 시뮬레이션을 통해 제시하였다.

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.

The paper is to solve the problem of existing audio signal transfer system which has a difficulties of system organization and the increase of additional install cost and unfriendly interior. To solve the existing system, we drew the new audio signal transfer system based on PLC and evaluated it. A transmitter and a receiver were designed using the PLC chip INT5500CS. An audio signal transfer system was configured with a CD player to which audio signals are sent from the transmitter and a speaker connected to the receiver. For performance evaluation of this system, a USBPre external sound card and Smaart Live 5 which is a PC-based sound measuring program were added. As a result of our experiment, the measured signal level is $2{\sim}3$ dB lower than reference signal, latency is 16.69 ms, and the specific character of coherency is bad in high frequency band. Otherwise, this system transmits and receives signals over 90 % in good condition as a result of measuring pink noise, frequency (1 kHz), and phase, magnitude. In view of the result so far achieved, the system designed this study has excellent performance, it resolves defect of existing audio signal transfer system.

$\textbullet$ Contents 1 Abstract $\textbullet$ Contents 2 Introduction $\textbullet$ Contents 3 Design of the system $\textbullet$ Contents 4 Creating product database $\textbullet$ Contents 5 Functionality testing $\textbullet$ Contents 6 Conclusion $\textbullet$ Contents 7 References

The rapid development of digital media and communication network urgently brings about the need of data certification technology to protect IPR (Intellectual property right). This paper proposed a new watermarking method for embedding contents owner's audio signal in order to protect color image IPR. Since this method evolves the existing static model and embeds audio signal of big data, it has the advantage of restoring signal transformed due to attacks. Three basic stages of watermarking include: 1) Encode analogue ID owner's audio signal using PCM and create new 3D audio watermark; 2) Interleave 3D audio watermark by linear bit expansion and 3) Transform Y signal of color image into wavelet and embed interleaved audio watermark in the low frequency band on the transform domain. The results demonstrated that the audio signal embedding in color image proposed in this paper enhanced robustness against lossy JPEG compression, standard image compression and image cropping and rotation which remove a part of image.

Various researches are being conducted on an integrated speech/audio coder capable of handling both speech and audio signals. However, since the speech and the audio signals have different characteristics, it is difficult to process them with a single coder. Therefore, instead of coding the speech and the audio with only one coder, a method of coding the speech with a speech coder and the audio with an audio coder was adopted, and an integrated speech/audio coder combining the speech and the audio coders was developed. That is, after signal classification is performed on the input signal, the speech is coded by the speech coder and the audio is coded by the audio coder. G.718 is a representative speech/audio coder that uses a signal classifier using modified discrete cosine transform (MDCT) energy ratio to classify input signals into the speech and the audio and then codes the input signals. However, since the signal classifier of G.718 performs a simple signal classification using a single parameter, the performance of the speech/audio signal classifier is not good, resulting in degradation of the overall performance of the speech/audio coder. Therefore, in this paper, a speech/audio signals classifier for the frame-by frame signal coding by a new decision rule is proposed. In the proposed classification, various time and frequency domain features are extracted. To design an optimized speech/audio classifier, the classification and regression tree (CART) algorithm is utilized and a binary decision tree is constructed. The performance of the proposed classifier with the various features is compared with that of the G.718 with the ratio between the MDCT power ratio of the input and the code-excited linear prediction (CELP) coder output signal. Experimental results show that the classification rates of the proposed algorithm are higher than those of the classifiers in G.718.

This paper describes a comprehensive process and range of design tools and components for providing Improved perception of engine sound for mass production vehicles by the generation of finely tuned engine harmonics.

감정과 관련된 다중 모달리티 신호의 심층 학습을 통한 상호보완적 정보 공유의 이점을 고려하여, 본 논문에서는 Electroencephalogram(EEG) 신호와 해당 오디오 신호를 기반으로 한 다중 모달리티 감정인식 방식을 제안한다. 제안하는 방식에서는 먼저 EEG 신경망(EEGNet)과 Temporal Convolution Network(TCN) 신경망을 사용하여 뇌전도 신호로부터 시공간 특징을 추출하고, 사전학습된 VGG신경망과 TCN 신경망을 사용하여 해당 오디오 신호로부터 시공간 특징을 추출한 다음, Conformer 메커니즘을 적용하여 뇌파 및 해당 오디오 특징을 병합하여 모델링함으로써 4가지의 감정 상태를 인식하였다. 감정인식에 널리 사용되는 DEAP 데이터세트에서 실험을 수행한 결과, 제안된 방식이 EEG 신호만 사용한 이전 연구와 비교하여 향상된 정확도를 보여주었다.

Considering the benefits of complementary information sharing through deep learning of multi-modal signals related to emotions, this paper proposes a multimodal emotion recognition method based on audio and Electroencephalogram (EEG) data. In the proposed method, we first extract spatio-temporal features from EEG signals using EEG networks (EEGNet) and Temporal Convolution Network (TCN), extract spatio-temporal features from corresponding audio signals using pre-trained VGG networks and TCN. Then, the Conformer mechanism is applied to merge EEG and audio features to recognize four emotional states. Experimental results on the DEAP dataset, which is widely used for emotion recognition, showed that the proposed method showed improved accuracy compared to previous studies using only EEG signals.

본 논문은 자기부호화기를 이용한 음향신호 분리방법을 제안한다. 사용된 복층구조 신경망 자기부호화기는 입력 신호의 효율적인 표현방법을 자동으로 학습하며, 유사한 특징을 가지고 있는 요소신호들을 군집함으로써 다른 특징의 신호들을 분리할 수 있다. 시간영역과 주파수영역의 변이특성을 추출하기 위하여 단구간푸리에변환(Short-Time Fourier Transform, STFT)을 수행하였으며, 정해진 크기의 사각형 창을 모든 가능한 위치에 적용하여 얻은 단구간 주파수 스펙트럼을 자기부호화기의 입력으로 사용하였다. 자기부호화기의 부호노드들의 값을 이용하여 유사한 스펙트럼 창들을 군집하고, 이를 이용하여 원래의 음원들로 분리해 낼 수 있었다. 분리된 원음들은 원래의 입력신호의 특징을 확실히 나타내었으며, 기존의 비음수 행렬분해(Non-negative Matrix Factorization, NMF) 결과와 주파수 스펙트럼 비교를 통해 그 유효성을 보일 수 있었다.

This paper proposes a novel approach to the problem of audio signal clustering using a stacked autoencoder. The proposed stacked autoencoder learns an efficient representation for the input signal, enables clustering constituent signals with similar characteristics, and therefore the original sources can be separated based on the clustering results. STFT (Short-Time Fourier Transform) is performed to extract time-frequency spectrum, and rectangular windows at all the possible locations are used as input values to the autoencoder. The outputs at the middle, encoding layer, are used to cluster the rectangular windows and the original sources are separated by the Wiener filters derived from the clustering results. Source separation experiments were carried out in comparison to the conventional NMF (Non-negative Matrix Factorization), and the estimated sources by the proposed method well represent the characteristics of the orignal sources as shown in the time-frequency representation.

We propose a novel scheme for digital audio authentication of given audio files which are edited by inserting small audio segments from different environmental sources. The purpose of this research is to detect inserted sections from given audio files. We expect that the proposed method will assist human investigators by notifying suspected audio section which considered to be recorded or transmitted on different environments. GMM-UBM and GSV-SVM are applied for modeling the dominant environment of a given audio file. Four kinds of likelihood ratio based scores and SVM score are used to measure the likelihood for a dominant environment model. We also use an ensemble score which is a combination of the aforementioned five kinds of scores. In the experimental results, the proposed method shows the lowest average equal error rate when we use the ensemble score. Even when dominant environments were unknown, the proposed method gives a similar accuracy.

본 논문은 오디오 부호화 및 복호화 과정에서, 주파수 영역에서 표현된 오디오 신호를 차원 축소 방법으로 압축하여 표현함으로서 오디오 부호화 효율을 증대시키고자 하는 방식에 관한 것이다. 차원 축소는 행렬을 특정한 조건을 바탕으로 두 개의 행렬의 곱으로 표현하는 방식으로, 특정 행렬로 표현된 데이터를 좀 더 작은 데이터량으로 표현하는 것뿐만 아니라 이 과정에서 데이터에 내재되어 있는 추상적인 정보까지도 함축적으로 얻어낼 수 있기 때문에, 일반적으로 데이터의 압축에 좋은 성능을 보인다. 주파수 영역으로 변환된 신호는 일반적으로 (주파수 밴드의 개수) $\times$ (전체 프레임의 개수)인 행렬로 볼 수 있으며, 이 전체 행렬을 입력으로 간주하고, 차원 축소를 수행하여 신호의 압축 효과를 얻을 수 있다. 그러나 이 경우, 행렬 전체를 입력 신호로 보아야 하기 때문에 실시간 부호화가 불가능하며, 신호 전체 길이만큼의 부호화 지연이 발생한다. 이를 해소하기 위해, 본 논문에서는 특정 개수만큼의 프레임을 묶어서 여러 번의 차원 축소를 순차적으로 수행함으로써 부호화 지연을 최소화하는 방식을 제안한다.

This paper describes the design and fabrication of a signal processing integrated circuit required for the recording and playback of VTR audio signal. The integrated circuit was designed using 8\ulcorner design rule and its chip size is 2.5x2.5mm\ulcorner It was fabricated using SST bipolar standard process technology. The measurement analysis of the fabricated circuit proves the satisfactory DC characteristics and its proper audio signal processing funcstion.

본 논문에서는 방송용 오디오 기기가 갖는 다채널의 특성과 각 채널에 대한 다양한 신호처리 기능의 특성을 고려하여 다채널 디지털 오디오 신호 처리기의 구조를 제안하고 범용 DSP를 이용하여 실시간 병렬 처리 시스템을 구현하였다. 구현된 시스템은 32비트 부동수소점 DSP를 이용하였으며 스테레오 채널의 48KHz 표본화 주파수를 지원하고 20비트 해상도를 갖는 시스템이다. 다채널 디지털 오디오 신호 처리 시스템의 구조는 디지털 신호 처리 과정을 수행하는 디지털 오디오 데이터 처리 부분과 시스템을 제어하기 위한 제어 정보 처리 부분으로 제안하였다. 이러한 구조에 적합한 실시간 시스템을 구현하기 위해 전체 시스템은 4부분의 모듈로 구성된다.