626 research outputs found
Speech enhancement with frequency domain auto-regressive modeling
Speech applications in far-field real world settings often deal with signals
that are corrupted by reverberation. The task of dereverberation constitutes an
important step to improve the audible quality and to reduce the error rates in
applications like automatic speech recognition (ASR). We propose a unified
framework of speech dereverberation for improving the speech quality and the
ASR performance using the approach of envelope-carrier decomposition provided
by an autoregressive (AR) model. The AR model is applied in the frequency
domain of the sub-band speech signals to separate the envelope and carrier
parts. A novel neural architecture based on dual path long short term memory
(DPLSTM) model is proposed, which jointly enhances the sub-band envelope and
carrier components. The dereverberated envelope-carrier signals are modulated
and the sub-band signals are synthesized to reconstruct the audio signal back.
The DPLSTM model for dereverberation of envelope and carrier components also
allows the joint learning of the network weights for the down stream ASR task.
In the ASR tasks on the REVERB challenge dataset as well as on the VOiCES
dataset, we illustrate that the joint learning of speech dereverberation
network and the E2E ASR model yields significant performance improvements over
the baseline ASR system trained on log-mel spectrogram as well as other
benchmarks for dereverberation (average relative improvements of 10-24% over
the baseline system). The speech quality improvements, evaluated using
subjective listening tests, further highlight the improved quality of the
reconstructed audio.Comment: 10 page
Auditory processing-based features for improving speech recognition in adverse acoustic conditions
n/
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 Bayesian Network View on Acoustic Model-Based Techniques for Robust Speech Recognition
This article provides a unifying Bayesian network view on various approaches
for acoustic model adaptation, missing feature, and uncertainty decoding that
are well-known in the literature of robust automatic speech recognition. The
representatives of these classes can often be deduced from a Bayesian network
that extends the conventional hidden Markov models used in speech recognition.
These extensions, in turn, can in many cases be motivated from an underlying
observation model that relates clean and distorted feature vectors. By
converting the observation models into a Bayesian network representation, we
formulate the corresponding compensation rules leading to a unified view on
known derivations as well as to new formulations for certain approaches. The
generic Bayesian perspective provided in this contribution thus highlights
structural differences and similarities between the analyzed approaches
- …