711 research outputs found
ORCA-SPOT: An Automatic Killer Whale Sound Detection Toolkit Using Deep Learning
Large bioacoustic archives of wild animals are an important source to identify reappearing communication patterns, which can then be related to recurring behavioral patterns to advance the current understanding of intra-specific communication of non-human animals. A main challenge remains that most large-scale bioacoustic archives contain only a small percentage of animal vocalizations and a large amount of environmental noise, which makes it extremely difficult to manually retrieve sufficient vocalizations for further analysis – particularly important for species with advanced social systems and complex vocalizations. In this study deep neural networks were trained on 11,509 killer whale (Orcinus orca) signals and 34,848 noise segments. The resulting toolkit ORCA-SPOT was tested on a large-scale bioacoustic repository – the Orchive – comprising roughly 19,000 hours of killer whale underwater recordings. An automated segmentation of the entire Orchive recordings (about 2.2 years) took approximately 8 days. It achieved a time-based precision or positive-predictive-value (PPV) of 93.2% and an area-under-the-curve (AUC) of 0.9523. This approach enables an automated annotation procedure of large bioacoustics databases to extract killer whale sounds, which are essential for subsequent identification of significant communication patterns. The code will be publicly available in October 2019 to support the application of deep learning to bioaoucstic research. ORCA-SPOT can be adapted to other animal species
Anonymizing Speech: Evaluating and Designing Speaker Anonymization Techniques
The growing use of voice user interfaces has led to a surge in the collection
and storage of speech data. While data collection allows for the development of
efficient tools powering most speech services, it also poses serious privacy
issues for users as centralized storage makes private personal speech data
vulnerable to cyber threats. With the increasing use of voice-based digital
assistants like Amazon's Alexa, Google's Home, and Apple's Siri, and with the
increasing ease with which personal speech data can be collected, the risk of
malicious use of voice-cloning and speaker/gender/pathological/etc. recognition
has increased.
This thesis proposes solutions for anonymizing speech and evaluating the
degree of the anonymization. In this work, anonymization refers to making
personal speech data unlinkable to an identity while maintaining the usefulness
(utility) of the speech signal (e.g., access to linguistic content). We start
by identifying several challenges that evaluation protocols need to consider to
evaluate the degree of privacy protection properly. We clarify how
anonymization systems must be configured for evaluation purposes and highlight
that many practical deployment configurations do not permit privacy evaluation.
Furthermore, we study and examine the most common voice conversion-based
anonymization system and identify its weak points before suggesting new methods
to overcome some limitations. We isolate all components of the anonymization
system to evaluate the degree of speaker PPI associated with each of them.
Then, we propose several transformation methods for each component to reduce as
much as possible speaker PPI while maintaining utility. We promote
anonymization algorithms based on quantization-based transformation as an
alternative to the most-used and well-known noise-based approach. Finally, we
endeavor a new attack method to invert anonymization.Comment: PhD Thesis Pierre Champion | Universit\'e de Lorraine - INRIA Nancy |
for associated source code, see https://github.com/deep-privacy/SA-toolki
Adaptation of speech recognition systems to selected real-world deployment conditions
Tato habilitační práce se zabývá problematikou adaptace systémů
rozpoznávání řeči na vybrané reálné podmínky nasazení. Je koncipována
jako sborník celkem dvanácti článků, které se touto problematikou
zabývají. Jde o publikace, jejichž jsem hlavním autorem
nebo spoluatorem, a které vznikly v rámci několika navazujících
výzkumných projektů. Na řešení těchto projektů jsem se
podílel jak v roli člena výzkumného týmu, tak i v roli řešitele nebo
spoluřešitele.
Publikace zařazené do tohoto sborníku lze rozdělit podle tématu
do tří hlavních skupin. Jejich společným jmenovatelem je
snaha přizpůsobit daný rozpoznávací systém novým podmínkám či
konkrétnímu faktoru, který významným způsobem ovlivňuje jeho
funkci či přesnost.
První skupina článků se zabývá úlohou neřízené adaptace na
mluvčího, kdy systém přizpůsobuje svoje parametry specifickým
hlasovým charakteristikám dané mluvící osoby. Druhá část práce
se pak věnuje problematice identifikace neřečových událostí na vstupu
do systému a související úloze rozpoznávání řeči s hlukem
(a zejména hudbou) na pozadí. Konečně třetí část práce se zabývá
přístupy, které umožňují přepis audio signálu obsahujícího promluvy
ve více než v jednom jazyce. Jde o metody adaptace existujícího
rozpoznávacího systému na nový jazyk a metody identifikace
jazyka z audio signálu.
Obě zmíněné identifikační úlohy jsou přitom vyšetřovány zejména
v náročném a méně probádaném režimu zpracování po jednotlivých
rámcích vstupního signálu, který je jako jediný vhodný pro on-line
nasazení, např. pro streamovaná data.This habilitation thesis deals with adaptation of automatic speech
recognition (ASR) systems to selected real-world deployment conditions.
It is presented in the form of a collection of twelve articles
dealing with this task; I am the main author or a co-author of these
articles. They were published during my work on several consecutive
research projects. I have participated in the solution of them
as a member of the research team as well as the investigator or a
co-investigator.
These articles can be divided into three main groups according to
their topics. They have in common the effort to adapt a particular
ASR system to a specific factor or deployment condition that affects
its function or accuracy.
The first group of articles is focused on an unsupervised speaker
adaptation task, where the ASR system adapts its parameters to
the specific voice characteristics of one particular speaker. The second
part deals with a) methods allowing the system to identify
non-speech events on the input, and b) the related task of recognition
of speech with non-speech events, particularly music, in the
background. Finally, the third part is devoted to the methods
that allow the transcription of an audio signal containing multilingual
utterances. It includes a) approaches for adapting the existing
recognition system to a new language and b) methods for identification
of the language from the audio signal.
The two mentioned identification tasks are in particular investigated
under the demanding and less explored frame-wise scenario,
which is the only one suitable for processing of on-line data streams
Deep Learning for Distant Speech Recognition
Deep learning is an emerging technology that is considered one of the most
promising directions for reaching higher levels of artificial intelligence.
Among the other achievements, building computers that understand speech
represents a crucial leap towards intelligent machines. Despite the great
efforts of the past decades, however, a natural and robust human-machine speech
interaction still appears to be out of reach, especially when users interact
with a distant microphone in noisy and reverberant environments. The latter
disturbances severely hamper the intelligibility of a speech signal, making
Distant Speech Recognition (DSR) one of the major open challenges in the field.
This thesis addresses the latter scenario and proposes some novel techniques,
architectures, and algorithms to improve the robustness of distant-talking
acoustic models. We first elaborate on methodologies for realistic data
contamination, with a particular emphasis on DNN training with simulated data.
We then investigate on approaches for better exploiting speech contexts,
proposing some original methodologies for both feed-forward and recurrent
neural networks. Lastly, inspired by the idea that cooperation across different
DNNs could be the key for counteracting the harmful effects of noise and
reverberation, we propose a novel deep learning paradigm called network of deep
neural networks. The analysis of the original concepts were based on extensive
experimental validations conducted on both real and simulated data, considering
different corpora, microphone configurations, environments, noisy conditions,
and ASR tasks.Comment: PhD Thesis Unitn, 201
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