9,863 research outputs found
Block-Online Multi-Channel Speech Enhancement Using DNN-Supported Relative Transfer Function Estimates
This work addresses the problem of block-online processing for multi-channel
speech enhancement. Such processing is vital in scenarios with moving speakers
and/or when very short utterances are processed, e.g., in voice assistant
scenarios. We consider several variants of a system that performs beamforming
supported by DNN-based voice activity detection (VAD) followed by
post-filtering. The speaker is targeted through estimating relative transfer
functions between microphones. Each block of the input signals is processed
independently in order to make the method applicable in highly dynamic
environments. Owing to the short length of the processed block, the statistics
required by the beamformer are estimated less precisely. The influence of this
inaccuracy is studied and compared to the processing regime when recordings are
treated as one block (batch processing). The experimental evaluation of the
proposed method is performed on large datasets of CHiME-4 and on another
dataset featuring moving target speaker. The experiments are evaluated in terms
of objective and perceptual criteria (such as signal-to-interference ratio
(SIR) or perceptual evaluation of speech quality (PESQ), respectively).
Moreover, word error rate (WER) achieved by a baseline automatic speech
recognition system is evaluated, for which the enhancement method serves as a
front-end solution. The results indicate that the proposed method is robust
with respect to short length of the processed block. Significant improvements
in terms of the criteria and WER are observed even for the block length of 250
ms.Comment: 10 pages, 8 figures, 4 tables. Modified version of the article
accepted for publication in IET Signal Processing journal. Original results
unchanged, additional experiments presented, refined discussion and
conclusion
A study of data coding technology developments in the 1980-1985 time frame, volume 2
The source parameters of digitized analog data are discussed. Different data compression schemes are outlined and analysis of their implementation are presented. Finally, bandwidth compression techniques are given for video signals
Kompetensi guru dalam pengajaran amali reka bentuk dan teknologi di Sekolah Rendah Daerah Batu Pahat
Kompetensi bermaksud kemampuan atau kecekapan seseorang individu dalam
melakukan sesuatu tugasan. Kompetensi juga merujuk kepada kemampuan
seseorang dalam melaksanakan sesuatu yang diperolehi melalui pendidikan dan
juga merujuk kepada prestasi dan perbuatan yang rasional untuk memenuhi
spesifikasi tertentu di dalam pelaksanaan tugas-tugas pendidikan. Objektif
kajian ini dijalankan adalah untuk mengenalpasti tahap kompetensi guru
terhadap pengajaran amali Reka Bentuk dan Teknologi di Sekolah Rendah
Daerah Batu Pahat. Kajian ini berbentuk tinjauan deskriptif yang menggunakan
borang soal selidik sebagai instrumen kajian. Borang soal selidik yang dibina
adalah berdasarkan kepada tiga elemen iaitu elemen pengetahuan, kemahiran
dan sikap. Seramai 118 orang guru yang mengajar mata pelajaran ini telah
terlibat sebagai responden. Data yang dikumpulkan telah dianalisis dengan
menggunakan perisian Statistical Package for Social Science (SPSS) versi 19
yang melibatkan statistik skor min dan ujian-T tidak bersandar. Hasil dapatan
kajian yang diperolehi menunjukkan guru-guru Reka Bentuk dan Teknoogi
mempunyai tahap kompetensi yang tinggi terhadap proses pengajaran amali
iaitu skor min yang diperolehi pada elemen pengetahuan adalah 4.23, elemen
kemahiran adalah 4.30, dan elemen sikap adalah 4.47. Dapatan kajian juga
menunjukkan tidak terdapat perbezaan yang signifikan terhadap tahap
kompetensi berdasarkan jantina guru lelaki dan guru perempuan dengan nilai
sigifikan melebihi 0.05 iaitu sebanyak 0.059. Beberapa cadangan untuk
penambahbaikan juga dikemukan dalam kajian ini. Hasil dari dapatan kajian ini
dapat digunakan sebagai cadangan garis panduan kepada guru-guru Reka
Bentuk dan Teknologi untuk mencapai Standard Kompetensi Guru
Self-Supervised Vision-Based Detection of the Active Speaker as Support for Socially-Aware Language Acquisition
This paper presents a self-supervised method for visual detection of the
active speaker in a multi-person spoken interaction scenario. Active speaker
detection is a fundamental prerequisite for any artificial cognitive system
attempting to acquire language in social settings. The proposed method is
intended to complement the acoustic detection of the active speaker, thus
improving the system robustness in noisy conditions. The method can detect an
arbitrary number of possibly overlapping active speakers based exclusively on
visual information about their face. Furthermore, the method does not rely on
external annotations, thus complying with cognitive development. Instead, the
method uses information from the auditory modality to support learning in the
visual domain. This paper reports an extensive evaluation of the proposed
method using a large multi-person face-to-face interaction dataset. The results
show good performance in a speaker dependent setting. However, in a speaker
independent setting the proposed method yields a significantly lower
performance. We believe that the proposed method represents an essential
component of any artificial cognitive system or robotic platform engaging in
social interactions.Comment: 10 pages, IEEE Transactions on Cognitive and Developmental System
NASA Tech Briefs Index, 1977, volume 2, numbers 1-4
Announcements of new technology derived from the research and development activities of NASA are presented. Abstracts, and indexes for subject, personal author, originating center, and Tech Brief number are presented for 1977
Robust sound event detection in bioacoustic sensor networks
Bioacoustic sensors, sometimes known as autonomous recording units (ARUs),
can record sounds of wildlife over long periods of time in scalable and
minimally invasive ways. Deriving per-species abundance estimates from these
sensors requires detection, classification, and quantification of animal
vocalizations as individual acoustic events. Yet, variability in ambient noise,
both over time and across sensors, hinders the reliability of current automated
systems for sound event detection (SED), such as convolutional neural networks
(CNN) in the time-frequency domain. In this article, we develop, benchmark, and
combine several machine listening techniques to improve the generalizability of
SED models across heterogeneous acoustic environments. As a case study, we
consider the problem of detecting avian flight calls from a ten-hour recording
of nocturnal bird migration, recorded by a network of six ARUs in the presence
of heterogeneous background noise. Starting from a CNN yielding
state-of-the-art accuracy on this task, we introduce two noise adaptation
techniques, respectively integrating short-term (60 milliseconds) and long-term
(30 minutes) context. First, we apply per-channel energy normalization (PCEN)
in the time-frequency domain, which applies short-term automatic gain control
to every subband in the mel-frequency spectrogram. Secondly, we replace the
last dense layer in the network by a context-adaptive neural network (CA-NN)
layer. Combining them yields state-of-the-art results that are unmatched by
artificial data augmentation alone. We release a pre-trained version of our
best performing system under the name of BirdVoxDetect, a ready-to-use detector
of avian flight calls in field recordings.Comment: 32 pages, in English. Submitted to PLOS ONE journal in February 2019;
revised August 2019; published October 201
Emotion Recognition from Acted and Spontaneous Speech
Dizertační práce se zabývá rozpoznáním emočního stavu mluvčích z řečového signálu. Práce je rozdělena do dvou hlavních častí, první část popisuju navržené metody pro rozpoznání emočního stavu z hraných databází. V rámci této části jsou představeny výsledky rozpoznání použitím dvou různých databází s různými jazyky. Hlavními přínosy této části je detailní analýza rozsáhlé škály různých příznaků získaných z řečového signálu, návrh nových klasifikačních architektur jako je například „emoční párování“ a návrh nové metody pro mapování diskrétních emočních stavů do dvou dimenzionálního prostoru. Druhá část se zabývá rozpoznáním emočních stavů z databáze spontánní řeči, která byla získána ze záznamů hovorů z reálných call center. Poznatky z analýzy a návrhu metod rozpoznání z hrané řeči byly využity pro návrh nového systému pro rozpoznání sedmi spontánních emočních stavů. Jádrem navrženého přístupu je komplexní klasifikační architektura založena na fúzi různých systémů. Práce se dále zabývá vlivem emočního stavu mluvčího na úspěšnosti rozpoznání pohlaví a návrhem systému pro automatickou detekci úspěšných hovorů v call centrech na základě analýzy parametrů dialogu mezi účastníky telefonních hovorů.Doctoral thesis deals with emotion recognition from speech signals. The thesis is divided into two main parts; the first part describes proposed approaches for emotion recognition using two different multilingual databases of acted emotional speech. The main contributions of this part are detailed analysis of a big set of acoustic features, new classification schemes for vocal emotion recognition such as “emotion coupling” and new method for mapping discrete emotions into two-dimensional space. The second part of this thesis is devoted to emotion recognition using multilingual databases of spontaneous emotional speech, which is based on telephone records obtained from real call centers. The knowledge gained from experiments with emotion recognition from acted speech was exploited to design a new approach for classifying seven emotional states. The core of the proposed approach is a complex classification architecture based on the fusion of different systems. The thesis also examines the influence of speaker’s emotional state on gender recognition performance and proposes system for automatic identification of successful phone calls in call center by means of dialogue features.
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