• Journal of Institute of Control, Robotics and Systems
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• v.13 no.4
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• pp.385-391
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• 2007

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era. Since the digital audio amplifier is based on the PWM signal processing, it is improper to analyze the principle of signal generation using linear system theories. In this paper, a class D digital audio amplifier based ADSM (Advanced Delta-Sigma Modulation) is considered. We first model the digital audio amplifier and then explain the operation principle using variable structure control algorithm. Moreover, the ripple signal generated by the hysteresis in the comparator has a significant effect on the system performance. Thus, we present a method to find the magnitude and the frequency of the ripple signal using describing function. Finally, simulations and experiments are provided to show the validity of the proposed methods.

• Journal of Broadcast Engineering
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• v.20 no.4
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• pp.632-640
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• 2015

This paper proposes a new method for fast graphic visualization of accurate frequency response of audio equalizer (EQ). When a logarithmic frequency scale is used, a frequency response in high resolution is required for accurate low-band frequency response. However, the high-resolution frequency response requires a huge amount of computational load, which makes the real-time graphic visualization of frequency response impossible. In order to solve the problem of computational load, the proposed method utilizes a low-resolution virtual frequency response in the mid band. It first computes the virtual frequency response of each filter of EQ in the mid band, and then moves it to the target band so that the result corresponds to the desired filter response. Then, it determines the final frequency response of EQ by combining all filter responses. The experiments confirm that the proposed method provides the frequency response of EQ which has an equivalent shape to that computed in high frequency resolution with huge computational load.

• The Journal of the Acoustical Society of Korea
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• v.39 no.3
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• pp.143-149
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• 2020

This paper presents the effect of the feature extraction methods used in the audio preprocessing on the classification performance of the Convolutional Neural Networks (CNN). We extract mel spectrogram, log mel spectrogram, Mel Frequency Cepstral Coefficient (MFCC), and delta MFCC from the UrbanSound8K dataset, which is widely used in environmental sound classification studies. Then we scale the data to 3 distributions. Using the data, we test four CNNs, VGG16, and MobileNetV2 networks for performance assessment according to the audio features and scaling. The highest recognition rate is achieved when using the unscaled log mel spectrum as the audio features. Although this result is not appropriate for all audio recognition problems but is useful for classifying the environmental sounds included in the Urbansound8K.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.232-237
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• 2013

The role of the window length of time-frequency transformation is important for the audio data hiding methods utilizing phase information. In this paper, the experiments for our audio data hiding method were conducted in order to evaluate the audio quality and robustness against reverberant environment. The experimental results showed the tendency that the worse audio quality but better robustness were obtained when the lengthy window was applied. The important reason for quality degradation was pre-echo which flatters the percussive sound. The results also indicated that the wireless communication theory related to the length of time-frequency transform can be applied in the field of audio data hiding and acoustic data transmission.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.367-374
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• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• Proceedings of the Acoustical Society of Korea Conference
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• 1984.12a
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• pp.86-90
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• 1984

In this paper, a method measuring the reverberation time simultaneously on all of the audio frequency range was studied. The system developed in this study is composed of a pink noise generator, filter bank, and microcomputer with graphic display. In the experiment, reverbration time data measured by existing analog method and by the system were compared, and were almost same through the audio frequency. Based on the conclusion, the subject method has more convenience and accuracy with algorithm program development without existing problems and it was found out that this kind of method is widely available for many branches of room accoustics and architectual acoustics.

• Journal of Advanced Marine Engineering and Technology
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• v.36 no.1
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• pp.166-171
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• 2012

In this paper, short-range underwater wireless communication technique using visible LEDs is proposed. As an alternative to conventional acoustic system, visible LED communications show high quality and high speed data transmission characteristics. We design a frequency modulation-based optical wireless audio transmission system. The CD4046B phase-locked loop device is applied to implement the frequency modulation and demodulation. With a transmission modulation of 100 kHz, audio signal has successfully received at a transmission distance of 30 cm.

• Journal of Biomedical Engineering Research
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• v.20 no.3
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• pp.275-281
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• 1999

Conventional PW Doppler systems suffer from the ambiguity of measured blood velocities due to the spectrum aliasing when the corresponding Doppler frequencies are greater than the Nyquist frequency. Base-line shift is a customary method for dealiasing the Doppler spectrums. I lowever, Doppler audio signals still remain unchanged even when the base-line shift method is applied. This paper de scribes an method for dealiasing both the Doppler spectra and audio signals by using sampling rate expansion, frequency shifting, and filtering poerations. For undirectional flows, the method can increase the maximum detectable Doppler frequency from the Nyquist limit of one-half of the Pulse Repetition Frequency(PRF) to the PRF. Experiments with real data have been performed to verify the proposed method.

• 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.

• Proceedings of the KIEE Conference
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• 2005.05a
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• pp.170-172
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• 2005

The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, use 1 bit representation with a sampling frequency of 2.8224 MHz (64 $\times$ 44.1 kHz). For multi-rate PCM (Pulse Code Modulation) input like as 48/96/192 kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital filter structure composed of sample-rate converter and interpolation filter for the DSD encoder with multi-rate (48/96/192 kHz) PCM input. without a external sample-rate converter.