95 research outputs found
Non-native children speech recognition through transfer learning
This work deals with non-native children's speech and investigates both
multi-task and transfer learning approaches to adapt a multi-language Deep
Neural Network (DNN) to speakers, specifically children, learning a foreign
language. The application scenario is characterized by young students learning
English and German and reading sentences in these second-languages, as well as
in their mother language. The paper analyzes and discusses techniques for
training effective DNN-based acoustic models starting from children native
speech and performing adaptation with limited non-native audio material. A
multi-lingual model is adopted as baseline, where a common phonetic lexicon,
defined in terms of the units of the International Phonetic Alphabet (IPA), is
shared across the three languages at hand (Italian, German and English); DNN
adaptation methods based on transfer learning are evaluated on significant
non-native evaluation sets. Results show that the resulting non-native models
allow a significant improvement with respect to a mono-lingual system adapted
to speakers of the target language
DNN adaptation by automatic quality estimation of ASR hypotheses
In this paper we propose to exploit the automatic Quality Estimation (QE) of
ASR hypotheses to perform the unsupervised adaptation of a deep neural network
modeling acoustic probabilities. Our hypothesis is that significant
improvements can be achieved by: i)automatically transcribing the evaluation
data we are currently trying to recognise, and ii) selecting from it a subset
of "good quality" instances based on the word error rate (WER) scores predicted
by a QE component. To validate this hypothesis, we run several experiments on
the evaluation data sets released for the CHiME-3 challenge. First, we operate
in oracle conditions in which manual transcriptions of the evaluation data are
available, thus allowing us to compute the "true" sentence WER. In this
scenario, we perform the adaptation with variable amounts of data, which are
characterised by different levels of quality. Then, we move to realistic
conditions in which the manual transcriptions of the evaluation data are not
available. In this case, the adaptation is performed on data selected according
to the WER scores "predicted" by a QE component. Our results indicate that: i)
QE predictions allow us to closely approximate the adaptation results obtained
in oracle conditions, and ii) the overall ASR performance based on the proposed
QE-driven adaptation method is significantly better than the strong, most
recent, CHiME-3 baseline.Comment: Computer Speech & Language December 201
Automatic Quality Estimation for ASR System Combination
Recognizer Output Voting Error Reduction (ROVER) has been widely used for
system combination in automatic speech recognition (ASR). In order to select
the most appropriate words to insert at each position in the output
transcriptions, some ROVER extensions rely on critical information such as
confidence scores and other ASR decoder features. This information, which is
not always available, highly depends on the decoding process and sometimes
tends to over estimate the real quality of the recognized words. In this paper
we propose a novel variant of ROVER that takes advantage of ASR quality
estimation (QE) for ranking the transcriptions at "segment level" instead of:
i) relying on confidence scores, or ii) feeding ROVER with randomly ordered
hypotheses. We first introduce an effective set of features to compensate for
the absence of ASR decoder information. Then, we apply QE techniques to perform
accurate hypothesis ranking at segment-level before starting the fusion
process. The evaluation is carried out on two different tasks, in which we
respectively combine hypotheses coming from independent ASR systems and
multi-microphone recordings. In both tasks, it is assumed that the ASR decoder
information is not available. The proposed approach significantly outperforms
standard ROVER and it is competitive with two strong oracles that e xploit
prior knowledge about the real quality of the hypotheses to be combined.
Compared to standard ROVER, the abs olute WER improvements in the two
evaluation scenarios range from 0.5% to 7.3%
Auditory processing-based features for improving speech recognition in adverse acoustic conditions
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Learning to Rank Microphones for Distant Speech Recognition
Fully exploiting ad-hoc microphone networks for distant speech recognition is
still an open issue. Empirical evidence shows that being able to select the
best microphone leads to significant improvements in recognition without any
additional effort on front-end processing. Current channel selection techniques
either rely on signal, decoder or posterior-based features. Signal-based
features are inexpensive to compute but do not always correlate with
recognition performance. Instead decoder and posterior-based features exhibit
better correlation but require substantial computational resources. In this
work, we tackle the channel selection problem by proposing MicRank, a learning
to rank framework where a neural network is trained to rank the available
channels using directly the recognition performance on the training set. The
proposed approach is agnostic with respect to the array geometry and type of
recognition back-end. We investigate different learning to rank strategies
using a synthetic dataset developed on purpose and the CHiME-6 data. Results
show that the proposed approach is able to considerably improve over previous
selection techniques, reaching comparable and in some instances better
performance than oracle signal-based measures
Robust Automatic Speech Recognition through On-line Semi Blind Signal Extraction
This paper describes the system used to process the data of the
CHiME Pascal 2011 competition, whose goal is to separate the
desired speech and recognize the commands being spoken. The
binaural recorded mixtures are processed by an on-line Semi-
Blind Source Extraction algorithm. The algorithm is based on
a multi-stage architecture combining the advantages of con-
strained Independent Component Analysis and Wiener-based
processing, allowing the estimation of the target signal with lim-
ited distortion. The recovered target signal is then fed to the rec-
ognizer which uses noise robust features based on Gammatone
Frequency Cepstral Coefficients. Moreover, model adaptation
to actual processing is applied as a further stage to reduce the
acoustic mismatch. Performance comparison between differ-
ent model/algorithmic settings is reported for both development
and test data sets
Automatic assessment of spoken language proficiency of non-native children
This paper describes technology developed to automatically grade Italian
students (ages 9-16) on their English and German spoken language proficiency.
The students' spoken answers are first transcribed by an automatic speech
recognition (ASR) system and then scored using a feedforward neural network
(NN) that processes features extracted from the automatic transcriptions.
In-domain acoustic models, employing deep neural networks (DNNs), are derived
by adapting the parameters of an original out of domain DNN
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