27,934 research outputs found
Band-pass filtering of the time sequences of spectral parameters for robust wireless speech recognition
In this paper we address the problem of automatic speech recognition when wireless speech communication systems are involved. In this context, three main sources of distortion should be considered: acoustic environment, speech coding and transmission errors. Whilst the first one has already received a lot of attention, the last two deserve further investigation in our opinion. We have found out that band-pass filtering of the recognition features improves ASR performance when distortions due to these particular communication systems are present. Furthermore, we have evaluated two alternative configurations at different bit error rates (BER) typical of these channels: band-pass filtering the LP-MFCC parameters or a modification of the RASTA-PLP using a sharper low-pass section perform consistently better than LP-MFCC and RASTA-PLP, respectively.Publicad
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
Efficient Implementation of the Room Simulator for Training Deep Neural Network Acoustic Models
In this paper, we describe how to efficiently implement an acoustic room
simulator to generate large-scale simulated data for training deep neural
networks. Even though Google Room Simulator in [1] was shown to be quite
effective in reducing the Word Error Rates (WERs) for far-field applications by
generating simulated far-field training sets, it requires a very large number
of Fast Fourier Transforms (FFTs) of large size. Room Simulator in [1] used
approximately 80 percent of Central Processing Unit (CPU) usage in our CPU +
Graphics Processing Unit (GPU) training architecture [2]. In this work, we
implement an efficient OverLap Addition (OLA) based filtering using the
open-source FFTW3 library. Further, we investigate the effects of the Room
Impulse Response (RIR) lengths. Experimentally, we conclude that we can cut the
tail portions of RIRs whose power is less than 20 dB below the maximum power
without sacrificing the speech recognition accuracy. However, we observe that
cutting RIR tail more than this threshold harms the speech recognition accuracy
for rerecorded test sets. Using these approaches, we were able to reduce CPU
usage for the room simulator portion down to 9.69 percent in CPU/GPU training
architecture. Profiling result shows that we obtain 22.4 times speed-up on a
single machine and 37.3 times speed up on Google's distributed training
infrastructure.Comment: Published at INTERSPEECH 2018.
(https://www.isca-speech.org/archive/Interspeech_2018/abstracts/2566.html
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