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Chord-based stepwise Korean Trot music generation technique using RNN-GAN 基于RNN-GAN的基于和弦的渐进式韩国快步音乐生成技术
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-11-01 DOI: 10.7776/ASK.2020.39.6.622
Seorim Hwang
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引用次数: 0
Aerodynamic noise reduction of fan motor unit of cordless vacuum cleaner by optimal designing of splitter blades for impeller 基于叶轮分流叶片优化设计的无绳吸尘器风机电机组气动降噪
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-11-01 DOI: 10.7776/ASK.2020.39.6.524
Kunwoo Kim, Seo-Yoon Ryu, C. Cheong, Seongjin Seo, Cheolmin Jang, Hanshin Seol
: In this study, noise radiated from a high-speed fan-motor unit for a cordless vacuum cleaner is reduced by designing splitter blades on the existing impeller. First of all, in order to investigate the flow field through a fan-motor unit, especially impeller, the unsteady incompressible Reynolds-Averaged Navier-Stokes (RANS) equations are numerically solved by using computational fluid dynamic technique. With predicted flow field results as input, the Ffowcs Williams-Hawkings (FW-H) integral equation is solved to predict aerodynamic noise radiated from the impeller. The validity of the numerical methods is confirmed by comparing the predicted sound pressure spectrum with the measured one. Further analysis of the predicted flow field shows that the strong vortex is formed between the impeller blades. As the vortex induces the loss of the flow field and acts as an aerodynamic noise source, supplementary splitter blades are designed to the existing impeller to suppress the identified vortex. The length and position of splitter are selected as design factors and the effect of each design factor on aerodynamic noise is numerically analyzed by using the Taguchi method. From this results, the optimum location and length of splitter for minimum radiated noise is determined. The finally selected design shows lower noise than the existing one.
在本研究中,通过在现有的叶轮上设计分流叶片,降低了高速无绳吸尘器风扇电机单元的噪声。首先,利用计算流体动力学技术对非定常不可压缩雷诺-平均纳维-斯托克斯(RANS)方程进行数值求解,研究了风机-电机单元特别是叶轮内部的流场。以预测流场结果为输入,求解Ffowcs williams - hawkins (FW-H)积分方程,预测叶轮辐射的气动噪声。通过与实测声压谱的比较,验证了数值方法的有效性。对预测流场的进一步分析表明,叶轮叶片之间形成了强涡。由于涡流会引起流场损失并成为气动噪声源,因此在现有叶轮上设计补充分流叶片来抑制识别出的涡流。选取分离器的长度和位置作为设计因素,采用田口法数值分析了各设计因素对气动噪声的影响。根据实验结果,确定了最小辐射噪声条件下分路器的最佳位置和长度。最终选择的设计比现有的设计具有更低的噪声。
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引用次数: 0
Size estimation of Sperm Whale in the East Sea of Korea using click signals 利用点击信号估算韩国东海抹香鲸的体型
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-11-01 DOI: 10.7776/ASK.2020.39.6.533
Young Geul Yoon, Kang-Hoon Choi, Dong-Gyun Han, Hawsun Sohn, J. Choi
A total length of sperm whales can be estimated by measuring the Inter-Pulse Interval(IPI) of their clicks composed by multiple pulses. The IPI is caused by the two-way travel time of the sound transmission in the spermaceti within the whale head. Therefore, the IPI can be used to measure the whale’s total length based on allometric relationships between head and body length. In this paper, the click signals recorded in the East Sea, Korea in 2017 were analyzed to estimate the size of sperm whales. The size of sperm whales calculated by the relationship between IPI and body length was 9.9 m to 10.9 m, which is corresponding to the size of an adult female or a juvenile male sperm whale. This non-lethal acoustic method has been demonstrated to accurately estimate the sperm whale size, and can provide useful information for domestic sperm whale monitoring.
抹香鲸的总长度可以通过测量由多个脉冲组成的咔嚓声的脉冲间隔(IPI)来估计。IPI是由声音在鲸脑内的双向传播时间引起的。因此,IPI可以用来测量鲸鱼的总长度基于头部和身体长度之间的异速关系。本文分析了2017年在韩国东海记录的“咔哒”声信号,并推测了抹香鲸的大小。通过IPI与体长的关系计算出的抹香鲸体型为9.9 ~ 10.9 m,对应于成年雌抹香鲸或幼年雄抹香鲸的体型。这种非致命的声学方法已经被证明可以准确地估计抹香鲸的大小,并且可以为国内的抹香鲸监测提供有用的信息。
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引用次数: 0
Sound event detection based on multi-channel multi-scale neural networks for home monitoring system used by the hard-of-hearing 基于多通道多尺度神经网络的听力困难家庭监测系统声事件检测
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-11-01 DOI: 10.7776/ASK.2020.39.6.600
Gi Yong Lee and Hyoung-Gook Kim
: In this paper, we propose a sound event detection method using a multi-channel multi-scale neural networks for sound sensing home monitoring for the hearing impaired. In the proposed system, two channels with high signal quality are selected from several wireless microphone sensors in home. The three features (time difference of arrival, pitch range, and outputs obtained by applying multi-scale convolutional neural network to log mel spectrogram) extracted from the sensor signals are applied to a classifier based on a bidirectional gated recurrent neural network to further improve the performance of sound event detection. The detected sound event result is converted into text along with the sensor position of the selected channel and provided to the hearing impaired. The experimental results show that the sound event detection method of the proposed system is superior to the existing method and can effectively deliver sound information to the hearing impaired
:在本文中,我们提出了一种使用多通道多尺度神经网络的声音事件检测方法,用于听力受损者的声音传感家庭监测。在所提出的系统中,从家中的几个无线麦克风传感器中选择两个具有高信号质量的通道。从传感器信号中提取的三个特征(到达时间差、音高范围和通过将多尺度卷积神经网络应用于对数mel频谱图获得的输出)被应用于基于双向门控递归神经网络的分类器,以进一步提高声音事件检测的性能。检测到的声音事件结果与所选通道的传感器位置一起被转换成文本,并被提供给听力受损者。实验结果表明,该系统的声音事件检测方法优于现有方法,能够有效地向听力受损者传递声音信息
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引用次数: 0
Multi-site based earthquake event classification using graph convolution networks 基于图卷积网络的多点地震事件分类
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-11-01 DOI: 10.7776/ASK.2020.39.6.615
Gwantae Kim, Bonhwa Ku, Hanseok Ko
In this paper, we propose a multi-site based earthquake event classification method using graph convolution networks. In the traditional earthquake event classification methods using deep learning, they used single-site observation to estimate seismic event class. However, to achieve robust and accurate earthquake event classification on the seismic observation network, the method using the information from the multi-site observations is needed, instead of using only single-site data. Firstly, our proposed model employs convolution neural networks to extract informative embedding features from the single-site observation. Secondly, graph convolution networks are used to integrate the features from several stations. To evaluate our model, we explore the model structure and the number of stations for ablation study. Finally, our multi-site based model outperforms up to 10 % accuracy and event recall rate compared to single-site based model.
本文提出了一种基于图卷积网络的多站点地震事件分类方法。在传统的基于深度学习的地震事件分类方法中,他们使用单点观测来估计地震事件的类别。然而,为了在地震观测台网上实现稳健、准确的地震事件分类,需要利用多站点观测信息的方法,而不是只利用单站点数据。首先,我们提出的模型采用卷积神经网络从单点观测中提取信息嵌入特征。其次,利用图卷积网络对多个站点的特征进行整合;为了评估我们的模型,我们探讨了模型的结构和烧蚀研究的台站数量。最后,与基于单站点的模型相比,我们基于多站点的模型的准确率和事件召回率高达10%。
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引用次数: 1
Own-ship noise cancelling method for towed line array sonars using a beam-formed reference signal 基于波束形成参考信号的拖曳线阵声呐自船降噪方法
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-11-01 DOI: 10.7776/ASK.2020.39.6.559
Danbi Lee
: This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm’s limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.
针对拖曳阵声呐的噪声问题,提出了一种消噪算法。利用声阵列的部分通道进行额外的波束形成,以获得对噪声轴承具有鲁棒性的参考波束信号。在参考波束的基础上,采用归一化最小均方(NLMS)算法进行频域自适应降噪。根据参考波束和输入波束信号的相干性估计了本船噪声的方位。本船噪声级是用估计的转向角的噪声波束模式来计算的,通过确定是否更新滤波器,使去除的信号电平不超过估计的噪声电平,从而防止目标信号的丢失。仿真结果表明,该算法在舰船偏离航向超过常规算法极限40%的情况下仍能保持良好的性能,即使目标信号的频率与舰船信号的频率相同,也能检测到目标。
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引用次数: 1
Dilated convolution and gated linear unit based sound event detection and tagging algorithm using weak label 基于扩展卷积和门控线性单元的弱标签声音事件检测与标记算法
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-09-01 DOI: 10.7776/ASK.2020.39.5.414
C. Park, DongHyun Kim, Hanseok Ko
In this paper, we propose a Dilated Convolution Gate Linear Unit (DCGLU) to mitigate the lack of sparsity and small receptive field problems caused by the segmentation map extraction process in sound event detection with weak labels. In the advent of deep learning framework, segmentation map extraction approaches have shown improved performance in noisy environments. However, these methods are forced to maintain the size of the feature map to extract the segmentation map as the model would be constructed without a pooling operation. As a result, the performance of these methods is deteriorated with a lack of sparsity and a small receptive field. To mitigate these problems, we utilize GLU to control the flow of information and Dilated Convolutional Neural Networks (DCNNs) to increase the receptive field without additional learning parameters. For the performance evaluation, we employ a URBAN-SED and self-organized bird sound dataset. The relevant experiments show that our proposed DCGLU model outperforms over other baselines. In particular, our method is shown to exhibit robustness against nature sound noises with three Signal to Noise Ratio (SNR) levels (20 dB, 10 dB and 0 dB).
在本文中,我们提出了一种扩展卷积门线性单元(DCGLU)来缓解弱标签声事件检测中由于分割图提取过程中缺乏稀疏性和小感受野问题。随着深度学习框架的出现,分割图提取方法在噪声环境中表现出了更好的性能。然而,这些方法必须保持特征映射的大小来提取分割映射,因为模型将在没有池化操作的情况下构建。结果,这些方法的性能下降,缺乏稀疏性和小的接受域。为了缓解这些问题,我们使用GLU来控制信息流,并使用扩展卷积神经网络(DCNNs)来增加接受野,而无需额外的学习参数。为了进行性能评估,我们使用了URBAN-SED和自组织的鸟类声音数据集。相关实验表明,我们提出的DCGLU模型优于其他基准。特别是,我们的方法对三种信噪比(SNR)水平(20 dB, 10 dB和0 dB)的自然声音噪声具有鲁棒性。
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引用次数: 1
A study on speech disentanglement framework based on adversarial learning for speaker recognition 基于对抗性学习的说话人识别语音解纠缠框架研究
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-09-01 DOI: 10.7776/ASK.2020.39.5.447
Yoohwan Kwon, Soo-Whan Chung, Hong-Goo Kang
In this paper, we propose a system to extract effective speaker representations from a speech signal using a deep learning method. Based on the fact that speech signal contains identity unrelated information such as text content, emotion, background noise, and so on, we perform a training such that the extracted features only represent speaker-related information but do not represent speaker-unrelated information. Specifically, we propose an auto-encoder based disentanglement method that outputs both speaker-related and speaker-unrelated embeddings using effective loss functions. To further improve the reconstruction performance in the decoding process, we also introduce a discriminator popularly used in Generative Adversarial Network (GAN) structure. Since improving the decoding capability is helpful for preserving speaker information and disentanglement, it results in the improvement of speaker verification performance. Experimental results demonstrate the effectiveness of our proposed method by improving Equal Error Rate (EER) on benchmark dataset, Voxceleb1.
在本文中,我们提出了一种使用深度学习方法从语音信号中提取有效说话人表示的系统。基于语音信号包含身份无关信息(如文本内容、情绪、背景噪声等)的事实,我们执行训练,使得提取的特征仅表示说话者相关信息,而不表示说话者无关信息。具体来说,我们提出了一种基于自动编码器的解纠缠方法,该方法使用有效损失函数输出与说话者相关和与说话者无关的嵌入。为了进一步提高解码过程中的重构性能,我们还介绍了一种在生成对抗性网络(GAN)结构中广泛使用的鉴别器。由于提高解码能力有助于保存说话人信息和解纠缠,因此提高了说话人验证性能。实验结果证明了我们提出的方法在基准数据集Voxceleb1上改进等误码率(EER)的有效性。
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引用次数: 0
Moored measurement of the ambient noise and analysis with environmental factors in the coastal sea of Jeju Island 济州岛近海环境噪声的系泊测量及环境因素分析
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-09-01 DOI: 10.7776/ASK.2020.39.5.390
Inyong Jeong, Soohong Min, D. Paeng
Underwater ambient noise was measured at the eastern and western costal sites of Jeju Island where the water depth was 20 m by a hydrophone moored at mid-depth (10 m) for 4 months. These eastern and western sites were selected as potential sites for offshore wind power generator and the current wave energy generator, respectively. Ambient noise was affected by environmental data such as wind and wave, which were collected from nearby weather stations and an observation station. Below 100 Hz, ambient noise was changed about 5 dB ~ 20 dB due to low and high tide. Below 1 kHz, wave and wind effects were the main source for ambient noise, varying up to 25 dB. Ambient noise was strongly influenced by wave at lower frequency and by wind at higher frequency up to over 1 kHz. The higher frequency range over 10 kHz was influenced by rainfall and biological sources, and the spectrum was measured about 10 dB higher than the peak spectrum level from Wenz curve at this frequency range.
在济州岛东部和西部沿海地区,通过在中深度(10米)系泊4个月的水听器测量了水深20米的水下环境噪声。这些东部和西部场地分别被选为海上风力发电机和当前波浪能发电机的潜在场地。环境噪声受到风和波浪等环境数据的影响,这些数据是从附近的气象站和观测站收集的。在100Hz以下,由于低潮和高潮,环境噪声变化约5dB~20dB。在1 kHz以下,波浪和风的影响是环境噪声的主要来源,变化幅度高达25 dB。环境噪声受到低频波和高达1kHz的风的强烈影响。10kHz以上的较高频率范围受到降雨和生物源的影响,在该频率范围内测得的频谱比Wenz曲线的峰值频谱水平高出约10dB。
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引用次数: 1
Temporal attention based animal sound classification 基于时间注意力的动物声音分类
IF 0.4 Q4 ACOUSTICS Pub Date : 2020-09-01 DOI: 10.7776/ASK.2020.39.5.406
Jungmin Kim, Young Lo Lee, Donghyeon Kim, Hanseok Ko
In this paper, to improve the classification accuracy of bird and amphibian acoustic sound, we utilize GLU (Gated Linear Unit) and Self-attention that encourages the network to extract important features from data and discriminate relevant important frames from all the input sequences for further performance improvement. To utilize acoustic data, we convert 1-D acoustic data to a log-Mel spectrogram. Subsequently, undesirable component such as background noise in the log-Mel spectrogram is reduced by GLU. Then, we employ the proposed temporal self-attention to improve classification accuracy. The data consist of 6-species of birds, 8-species of amphibians including endangered species in the natural environment. As a result, our proposed method is shown to achieve an accuracy of 91 % with bird data and 93 % with amphibian data. Overall, an improvement of about 6 % ~ 7 % accuracy in performance is achieved compared to the existing algorithms.
为了提高鸟类和两栖动物声音的分类精度,本文采用了门控线性单元(GLU)和自注意(Self-attention)方法,鼓励网络从数据中提取重要特征,并从所有输入序列中区分出相关的重要帧,从而进一步提高性能。为了利用声学数据,我们将一维声学数据转换为对数梅尔谱图。在此基础上,对对数mel谱图中的背景噪声等不良成分进行了滤波。然后,我们利用提出的时间自关注来提高分类精度。该数据包括6种鸟类,8种两栖动物,包括自然环境中的濒危物种。结果表明,我们提出的方法对鸟类数据的准确率为91%,对两栖动物数据的准确率为93%。总体而言,与现有算法相比,该算法的性能精度提高了6% ~ 7%。
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引用次数: 1
期刊
Journal of the Acoustical Society of Korea
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