12 research outputs found
Rank-1 Constrained Multichannel Wiener Filter for Speech Recognition in Noisy Environments
Multichannel linear filters, such as the Multichannel Wiener Filter (MWF) and
the Generalized Eigenvalue (GEV) beamformer are popular signal processing
techniques which can improve speech recognition performance. In this paper, we
present an experimental study on these linear filters in a specific speech
recognition task, namely the CHiME-4 challenge, which features real recordings
in multiple noisy environments. Specifically, the rank-1 MWF is employed for
noise reduction and a new constant residual noise power constraint is derived
which enhances the recognition performance. To fulfill the underlying rank-1
assumption, the speech covariance matrix is reconstructed based on eigenvectors
or generalized eigenvectors. Then the rank-1 constrained MWF is evaluated with
alternative multichannel linear filters under the same framework, which
involves a Bidirectional Long Short-Term Memory (BLSTM) network for mask
estimation. The proposed filter outperforms alternative ones, leading to a 40%
relative Word Error Rate (WER) reduction compared with the baseline Weighted
Delay and Sum (WDAS) beamformer on the real test set, and a 15% relative WER
reduction compared with the GEV-BAN method. The results also suggest that the
speech recognition accuracy correlates more with the Mel-frequency cepstral
coefficients (MFCC) feature variance than with the noise reduction or the
speech distortion level.Comment: for Computer Speech and Languag
Signal-Adaptive and Perceptually Optimized Sound Zones with Variable Span Trade-Off Filters
Creating sound zones has been an active research field since the idea was
first proposed. So far, most sound zone control methods rely on either an
optimization of physical metrics such as acoustic contrast and signal
distortion or a mode decomposition of the desired sound field. By using these
types of methods, approximately 15 dB of acoustic contrast between the
reproduced sound field in the target zone and its leakage to other zone(s) has
been reported in practical set-ups, but this is typically not high enough to
satisfy the people inside the zones. In this paper, we propose a sound zone
control method shaping the leakage errors so that they are as inaudible as
possible for a given acoustic contrast. The shaping of the leakage errors is
performed by taking the time-varying input signal characteristics and the human
auditory system into account when the loudspeaker control filters are
calculated. We show how this shaping can be performed using variable span
trade-off filters, and we show theoretically how these filters can be used for
trading signal distortion in the target zone for acoustic contrast. The
proposed method is evaluated based on physical metrics such as acoustic
contrast and perceptual metrics such as STOI. The computational complexity and
processing time of the proposed method for different system set-ups are also
investigated. Lastly, the results of a MUSHRA listening test are reported. The
test results show that the proposed method provides more than 20% perceptual
improvement compared to existing sound zone control methods.Comment: Accepted for publication in IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH,
AND LANGUAGE PROCESSIN