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2020 28th European Signal Processing Conference (EUSIPCO)最新文献

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Analysis of Phonetic Dependence of Segmentation Errors in Speaker Diarization 说话人分词错误的语音依赖性分析
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287552
Simon W. McKnight, Aidan O. T. Hogg, P. Naylor
Evaluation of speaker segmentation and diarization normally makes use of forgiveness collars around ground truth speaker segment boundaries such that estimated speaker segment boundaries with such collars are considered completely correct. This paper shows that the popular recent approach of removing forgiveness collars from speaker diarization evaluation tools can unfairly penalize speaker diarization systems that correctly estimate speaker segment boundaries. The uncertainty in identifying the start and/or end of a particular phoneme means that the ground truth segmentation is not perfectly accurate, and even trained human listeners are unable to identify phoneme boundaries with full consistency. This research analyses the phoneme dependence of this uncertainty, and shows that it depends on (i) whether the phoneme being detected is at the start or end of an utterance and (ii) what the phoneme is, so that the use of a uniform forgiveness collar is inadequate. This analysis is expected to point the way towards more indicative and repeatable assessment of the performance of speaker diarization systems.
说话人分割和分割的评估通常在真实说话人段边界周围使用宽恕圈,这样使用这种圈估计的说话人段边界被认为是完全正确的。本文表明,最近流行的从说话人分界评价工具中去除宽恕圈的方法可能会对正确估计说话人分界的说话人分界系统造成不公平的惩罚。识别特定音素的开始和/或结束的不确定性意味着基础真值分割并不完全准确,甚至训练有素的人类听众也无法完全一致地识别音素边界。本研究分析了这种不确定性的音素依赖性,并表明它取决于(i)被检测的音素是在话语的开始还是结束以及(ii)音素是什么,因此使用统一的宽恕项圈是不够的。这一分析有望为对扬声器拨号系统的性能进行更具指示性和可重复性的评估指明道路。
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引用次数: 3
Orientation-Matched Multiple Modeling for RSSI-based Indoor Localization via BLE Sensors 基于rssi的BLE传感器室内定位方向匹配多重建模
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287489
M. Atashi, Parvin Malekzadeh, Mohammad Salimibeni, Zohreh Hajiakhondi-Meybodi, K. Plataniotis, Arash Mohammadi
Internet of Things (IoT) has penetrated different aspects of our modern life where smart sensors enabled with Bluetooth Low Energy (BLE) are deployed increasingly within our surrounding indoor environments. BLE-based localization is, typically, performed based on Received Signal Strength Indicator (RSSI), which suffers from different drawbacks due to its significant fluctuations. In this paper, we focus on a multiplemodel estimation framework for analyzing and addressing effects of orientation of a BLE-enabled device on indoor localization accuracy. The fusion unit of the proposed method would merge orientation estimated by RSSI values and heading estimated by Inertial Measurement Unit (IMU) sensors to gain higher accuracy in orientation classification. In contrary to existing RSSIbased solutions that use a single path-loss model, the proposed framework consists of eight orientation-matched path loss models coupled with a multi-sensor and data-driven classification model that estimates the orientation of a hand-held device with high accuracy of 99%. By estimating the orientation, we could mitigate the effect of orientation on the RSSI values and consequently improve RSSI-based distance estimates. In particular, the proposed data-driven and multiple-model framework is constructed based on over 10 million RSSI values and IMU sensor data collected via an implemented LBS platform.
物联网(IoT)已经渗透到我们现代生活的各个方面,支持低功耗蓝牙(BLE)的智能传感器越来越多地部署在我们周围的室内环境中。基于ble的定位通常基于接收信号强度指标(Received Signal Strength Indicator, RSSI),但RSSI的波动较大,存在不同的缺点。在本文中,我们重点研究了一个多模型估计框架,用于分析和解决启用ble的设备的方向对室内定位精度的影响。该方法的融合单元将RSSI值估计的方向与惯性测量单元(IMU)传感器估计的航向进行融合,以获得更高的方向分类精度。与现有的基于rssi的解决方案使用单一路径损耗模型相反,该框架由八个方向匹配的路径损耗模型以及一个多传感器和数据驱动的分类模型组成,该模型可以估计手持设备的方向,准确率高达99%。通过对方向的估计,可以减轻方向对RSSI值的影响,从而提高基于RSSI的距离估计。特别是,所提出的数据驱动和多模型框架是基于通过实现的LBS平台收集的超过1000万个RSSI值和IMU传感器数据构建的。
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引用次数: 11
The Smoothed Reassigned Spectrogram for Robust Energy Estimation 用于鲁棒能量估计的平滑重分配谱图
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287396
Erik Månsson, Maria Sandsten
The matched window reassigned spectrogram relocates all signal energy of an oscillating transient to the time-and frequency locations, resulting in a sharp peak in the time-frequency plane. However, previous research has shown that the method may result in split energy peaks for close components and in high noise levels, and the peak energy is then erroneously estimated. With use of novel knowledge on the statistics when subjected to noise, we propose a novel method, the smoothed reassigned spectrogram, for obtaining a stable and accurate measure of the signal energy from the peak value, with retained resolution properties. We also suggest a simple set of rules to enhance the reassigned spectrogram and speed up its calculation. Simulations are performed to verify the accuracy and an application example on radar data is shown.
匹配的窗口重分配谱图将振荡瞬态信号的所有能量重新定位到时间和频率位置,从而在时间-频率平面上产生一个尖锐的峰值。然而,先前的研究表明,该方法可能会导致接近组件的分裂能量峰值和高噪声水平,然后错误地估计峰值能量。利用噪声统计方面的新知识,我们提出了一种新的方法,即平滑重分配谱图,用于从峰值获得稳定而准确的信号能量测量,并保留分辨率特性。我们还提出了一套简单的规则来增强重分配谱图并加快其计算速度。通过仿真验证了该方法的准确性,并给出了在雷达数据上的应用实例。
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引用次数: 1
Transfer learning from speech to music: towards language-sensitive emotion recognition models 从语音到音乐的迁移学习:对语言敏感的情感识别模型
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287548
Juan Sebastián Gómez Cañón, Estefanía Cano, P. Herrera, E. Gómez
In this study, we address emotion recognition using unsupervised feature learning from speech data, and test its transferability to music. Our approach is to pre-train models using speech in English and Mandarin, and then fine-tune them with excerpts of music labeled with categories of emotion. Our initial hypothesis is that features automatically learned from speech should be transferable to music. Namely, we expect the intra-linguistic setting (e.g., pre-training on speech in English and fine-tuning on music in English) should result in improved performance over the cross-linguistic setting (e.g., pre-training on speech in English and fine-tuning on music in Mandarin). Our results confirm previous research on cross-domain transferability, and encourage research towards language-sensitive Music Emotion Recognition (MER) models.
在本研究中,我们使用语音数据中的无监督特征学习来解决情感识别问题,并测试其对音乐的可转移性。我们的方法是使用英语和普通话语音对模型进行预训练,然后用标记有情感类别的音乐片段对模型进行微调。我们最初的假设是,从语音中自动学习到的特征应该可以转移到音乐中。也就是说,我们期望语言内设置(例如,英语语音的预训练和英语音乐的微调)应该比跨语言设置(例如,英语语音的预训练和中文音乐的微调)产生更好的性能。我们的研究结果证实了之前关于跨领域可转移性的研究,并鼓励对语言敏感的音乐情感识别(MER)模型的研究。
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引用次数: 1
The Modulo Radon Transform and its Inversion 模Radon变换及其反演
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287586
A. Bhandari, Matthias Beckmann, F. Krahmer
In this paper, we introduce the Modulo Radon Transform (MRT) which is complemented by an inversion algorithm. The MRT generalizes the conventional Radon Transform and is obtained via computing modulo of the line integral of a two-dimensional function at a given angle. Since the modulo operation has an aliasing effect on the range of a function, the recorded MRT sinograms are always bounded, thus avoiding information loss arising from saturation or clipping effects. This paves a new pathway for imaging applications such as high dynamic range tomography, a topic that is in its early stages of development. By capitalizing on the recent results on Unlimited Sensing architecture, we prove that the Modulo Radon Transform can be inverted when the resultant (discrete/continuous) measurements map to a band-limited function. Thus, the MRT leads to new possibilities for both conceptualization of inversion algorithms as well as development of new hardware, for instance, for single-shot high dynamic range tomography.
在本文中,我们引入了模Radon变换(MRT),并辅以一种反演算法。MRT是传统Radon变换的推广,通过计算二维函数在给定角度处的线积分的模得到。由于模操作对函数的范围有混叠效应,因此记录的MRT信号图总是有界的,从而避免了由饱和或剪切效应引起的信息丢失。这为成像应用铺平了一条新途径,如高动态范围断层扫描,这是一个处于早期发展阶段的主题。通过利用Unlimited Sensing架构的最新结果,我们证明了当结果(离散/连续)测量映射到带限制函数时,模Radon变换可以反转。因此,MRT为反演算法的概念化以及新硬件的开发带来了新的可能性,例如,用于单镜头高动态范围层析成像。
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引用次数: 21
Micro-Doppler Signal Representation for Drone Classification by Deep Learning 基于深度学习的无人机分类微多普勒信号表示
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287525
Julien Gérard, J. Tomasik, C. Morisseau, Arpad Rimmel, G. Vieillard
There are numerous formats which represent the micro-Doppler signature. Our goal is to determine which one is the most adapted to classify small UAV (Unmanned Aerial Vehicules) with Deep Learning. To achieve this goal, we compare drone classification results with the different micro-Doppler signatures for a given neural network. This comparison has been performed on data obtained during a radar measurement campaign. We evaluate the classification performance in function of different use conditions we identified with a given neural network. According to the experiments conducted, the recommended format is a spectrum issued from long observations as its classification results are better for most criteria.
有许多表示微多普勒信号的格式。我们的目标是确定哪一个最适合用深度学习对小型无人机(无人机)进行分类。为了实现这一目标,我们将无人机分类结果与给定神经网络的不同微多普勒特征进行比较。这种比较是在一次雷达测量活动中获得的数据上进行的。我们用给定的神经网络来评估我们识别的不同使用条件下的分类性能。根据所进行的实验,推荐的格式是由长期观察得出的光谱,因为它的分类结果对大多数标准都更好。
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引用次数: 10
Design of a Non-negative Neural Network to Improve on NMF 改进NMF的非负神经网络设计
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287668
Filip Wen-Fwu Tsai, Alireza M. Javid, S. Chatterjee
For prediction of a non-negative target signal using a non-negative input, we design a feed-forward neural network to achieve a better performance than a non-negative matrix factorization (NMF) algorithm. We provide a mathematical relation between the neural network and NMF. The architecture of the neural network is built on a property of rectified-linear-unit (ReLU) activation function and a convex optimization layer-wise training approach. For an illustrative example, we choose a speech enhancement application where a clean speech spectrum is estimated from a noisy spectrum.
为了使用非负输入预测非负目标信号,我们设计了一种前馈神经网络,以获得比非负矩阵分解(NMF)算法更好的性能。我们给出了神经网络和NMF之间的数学关系。神经网络的结构是建立在整流线性单元(ReLU)激活函数和凸优化分层训练方法的基础上的。作为一个说明性的例子,我们选择一个语音增强应用,其中从噪声频谱估计干净的语音频谱。
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引用次数: 0
Unbiased FIR Filtering under Bernoulli-Distributed Binary Randomly Delayed and Missing Data 伯努利分布二进制随机延迟和缺失数据下的无偏FIR滤波
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287509
Karen J. Uribe-Murcia, J. Andrade-Lucio, Y. Shmaliy, Yuan Xu
This paper develops an unbiased finite impulse response (UFIR) filtering algorithm for networked systems where uncertain delays and packet dropouts can happen due to measurement failures and unreliable communication. The binary Bernoulli distribution with known delay probability is used to model the randomly arrived measures. A novel representation of the stochastic model is presented for FIR-type filter structures. To avoid packet dropouts and improve the estimation accuracy when a message arrives with no data, a predictive algorithm is used. An advantage of the UFIR filtering approach is demonstrated by comparing the mean square errors with the Kalman and H∞ filters under the same conditions. Experimental verifications are provided based on GPS vehicle tracking.
针对网络系统中由于测量失败和通信不可靠而产生的不确定延迟和丢包,提出了一种无偏有限脉冲响应(UFIR)滤波算法。采用已知延迟概率的二元伯努利分布对随机到达测度进行建模。针对fir型滤波器结构,提出了一种新的随机模型表示。为了避免在没有数据的消息到达时丢包并提高估计精度,使用了预测算法。通过与卡尔曼滤波和H∞滤波在相同条件下的均方误差比较,证明了UFIR滤波方法的优点。给出了基于GPS车辆跟踪的实验验证。
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引用次数: 2
Full-Duplex mmWave Communication with Hybrid Precoding and Combining 基于混合预编码和组合的全双工毫米波通信
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287814
Roberto López-Valcarce, Marcos Martínez-Cotelo
We investigate the design of hybrid precoders and combiners for a millimeter wave (mmWave) point-to-point bidirectional link in which both nodes transmit and receive simultaneously and on the same carrier frequency. In such full-duplex configuration, mitigation of self-interference (SI) becomes critical. Large antenna arrays provide an opportunity for spatial SI suppression in mmWave. We assume a phase-shifter based, fully connected architecture for the analog part of the precoder and combiner. The proposed design, which aims at cancelling SI in the analog domain to avoid frontend saturation, significantly improves on the performance of previous approaches.
我们研究了用于毫米波(mmWave)点对点双向链路的混合预编码器和合并器的设计,其中两个节点同时在相同的载波频率上发送和接收。在这种全双工配置中,减轻自干扰(SI)变得至关重要。大型天线阵列为毫米波的空间SI抑制提供了机会。我们假设预编码器和组合器的模拟部分采用基于移相器的全连接架构。提出的设计旨在消除模拟域中的SI以避免前端饱和,显著提高了以前方法的性能。
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引用次数: 7
Exploiting Attention-based Sequence-to-Sequence Architectures for Sound Event Localization 利用基于注意力的序列到序列架构进行声音事件定位
Pub Date : 2021-01-24 DOI: 10.23919/Eusipco47968.2020.9287224
C. Schymura, Tsubasa Ochiai, Marc Delcroix, K. Kinoshita, T. Nakatani, S. Araki, D. Kolossa
Sound event localization frameworks based on deep neural networks have shown increased robustness with respect to reverberation and noise in comparison to classical parametric approaches. In particular, recurrent architectures that incorporate temporal context into the estimation process seem to be well-suited for this task. This paper proposes a novel approach to sound event localization by utilizing an attention-based sequence-to-sequence model. These types of models have been successfully applied to problems in natural language processing and automatic speech recognition. In this work, a multi-channel audio signal is encoded to a latent representation, which is subsequently decoded to a sequence of estimated directions-of-arrival. Herein, attentions allow for capturing temporal dependencies in the audio signal by focusing on specific frames that are relevant for estimating the activity and direction-of-arrival of sound events at the current time-step. The framework is evaluated on three publicly available datasets for sound event localization. It yields superior localization performance compared to state-of-the-art methods in both anechoic and reverberant conditions.
与经典参数方法相比,基于深度神经网络的声音事件定位框架在混响和噪声方面显示出更高的鲁棒性。特别是,将时间上下文合并到评估过程中的循环架构似乎非常适合这项任务。本文提出了一种基于注意力的序列到序列模型的声音事件定位方法。这些类型的模型已经成功地应用于自然语言处理和自动语音识别问题。在这项工作中,将多通道音频信号编码为潜在表示,随后将其解码为估计到达方向的序列。在此,通过关注与估计当前时间步长声音事件的活动和到达方向相关的特定帧,可以捕获音频信号中的时间依赖性。该框架在三个公开可用的声音事件定位数据集上进行了评估。与最先进的方法相比,它在消声和混响条件下都具有优越的定位性能。
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引用次数: 8
期刊
2020 28th European Signal Processing Conference (EUSIPCO)
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