515 research outputs found
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
Robust Speech Recognition Using Generative Adversarial Networks
This paper describes a general, scalable, end-to-end framework that uses the
generative adversarial network (GAN) objective to enable robust speech
recognition. Encoders trained with the proposed approach enjoy improved
invariance by learning to map noisy audio to the same embedding space as that
of clean audio. Unlike previous methods, the new framework does not rely on
domain expertise or simplifying assumptions as are often needed in signal
processing, and directly encourages robustness in a data-driven way. We show
the new approach improves simulated far-field speech recognition of vanilla
sequence-to-sequence models without specialized front-ends or preprocessing
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