Title: An Analysis of the Aurora Large Vocabulary Evaluation
1An Analysis of theAurora Large Vocabulary
Evaluation
EUROSPEECH 2003
- Authors
- Naveen Parihar and Joseph Picone
- Inst. for Signal and Info. Processing
- Dept. Electrical and Computer Eng.
- Mississippi State University
- Contact Information
- Box 9571
- Mississippi State University
- Mississippi State, Mississippi 39762
- Tel 662-325-8335
- Fax 662-325-2298
-
Email parihar,picone_at_isip.msstate.edu
URL isip.msstate.edu/publications/conferences/e
urospeech/2003/evaluation/
2INTRODUCTION
ABSTRACT
In this paper, we analyze the results of the
recent Aurora large vocabulary evaluations (ALV).
Two consortia submitted proposals on speech
recognition front ends for this evaluation
(1) Qualcomm, ICSI, and OGI (QIO), and
(2) Motorola, France Telecom, and Alcatel (MFA).
These front ends used a variety of noise
reduction techniques including discriminative
transforms, feature normalization, voice activity
detection, and blind equalization. Participants
used a common speech recognition engine to
post-process their features. In this paper, we
show that the results of this evaluation were not
significantly impacted by suboptimal recognition
system parameter settings. Without any front end
specific tuning, the MFA front end outperforms
the QIO front end by 9.6 relative. With tuning,
the relative performance gap increases to 15.8.
Both the mismatched microphone and additive noise
evaluation conditions resulted in a significant
degradation in performance for both front ends.
3INTRODUCTION
MOTIVATION
ALV Evaluation Results
- ALV goal was at least a 25 relative improvement
over the baseline MFCC front end - Two consortia participated
- QIO QualComm, ICSI, OGI
- MFA Motorola, France Telecom, Alcatel
- Generic baseline LVCSR system with no front end
specific tuning - Would front end specific tuning change the
rankings?
4EVALUATION PARADIGM
THE AURORA 4 DATABASE
- Acoustic Training
- Derived from 5000 word WSJ0 task
- TS1 (clean), and TS2 (multi-condition)
- Clean plus 6 noise conditions
- Randomly chosen SNR between 10 and 20 dB
- 2 microphone conditions (Sennheiser and
secondary) - 2 sample frequencies 16 kHz and 8 kHz
- G.712 filtering at 8 kHz and P.341 filtering at
16 kHz
- Development and Evaluation Sets
- Derived from WSJ0 Evaluation and Development sets
- 14 test sets for each
- 7 recorded on Sennheiser 7 on secondary
- Clean plus 6 noise conditions
- Randomly chosen SNR between 5 and 15 dB
- G.712 filtering at 8 kHz and P.341 filtering at
16 kHz
5EVALUATION PARADIGM
BASELINE LVCSR SYSTEM
- Standard context-dependent cross-word HMM-based
system - Acoustic models state-tied4-mixture cross-word
triphones - Language model WSJ0 5K bigram
- Search Viterbi one-best using lexical trees for
N-gram cross-word decoding - Lexicon based on CMUlex
- Real-time 4 xRT for training and 15 xRT for
decoding on an800 MHz Pentium
6EVALUATION PARADIGM
WI007 ETSI MFCC FRONT END
Input Speech
- The baseline HMM system used an ETSI standard
MFCC-based front end
Zero-mean and Pre-emphasis
- Zero-mean debiasing
- 10 ms frame duration
- 25 ms Hamming window
- Absolute energy
- 12 cepstral coefficients
- First and second derivatives
Fourier Transf. Analysis
Energy
Cepstral Analysis
7FRONT END PROPOSALS
QIO FRONT END
Input Speech
Qualcomm, ICSI, OGI (QIO) front end
Fourier Transform
- 10 msec frame duration
- 25 msec analysis window
- 15 RASTA-like filtered cepstral coefficients
- MLP-based VAD
- Mean and variance normalization
- First and second derivatives
Mel-scale Filter Bank
RASTA
MLP-based VAD
DCT
Mean/Variance Normalization
/
8FRONT END PROPOSALS
MFA FRONT END
- 10 msec frame duration
- 25 msec analysis window
- Mel-warped Wiener filter based noise reduction
- Energy-based VADNest
- Waveform processing to enhance SNR
- Weighted log-energy
- 12 cepstral coefficients
- Blind equalization (cepstral domain)
- VAD based on acceleration of various energy based
measures - First and second derivatives
9EXPERIMENTAL RESULTS
FRONT END SPECIFIC TUNING
- Pruning beams (word, phone and state) were opened
during the tuning process to eliminate search
errors. - Tuning parameters
- State-tying thresholds solves the problem of
sparsity of training data by sharing state
distributions among phonetically similar states - Language model scale controls influence of the
language model relative to the acoustic models
(more relevant for WSJ) - Word insertion penalty balances insertions and
deletions (always a concern in noisy environments)
10EXPERIMENTAL RESULTS
FRONT END SPECIFIC TUNING - QIO
- Parameter tuning
- clean data recorded on Sennhieser mic.
(corresponds to Training Set 1 and Devtest Set 1
of the Aurora-4 database) - 8 kHz sampling frequency
- 7.5 relative improvement
11EXPERIMENTAL RESULTS
FRONT END SPECIFIC TUNING - MFA
- Parameter tuning
- clean data recorded on Sennhieser mic.
(corresponds to Training Set 1 and Devtest Set 1
of the Aurora-4 database) - 8 kHz sampling frequency
- 9.4 relative improvement
- Ranking is still the same (14.9 vs. 12.5) !
12EXPERIMENTAL RESULTS
COMPARISON OF TUNING
- Same Ranking relative performance gap increased
from9.6 to 15.8 - On TS1, MFA FE significantly better on all 14
test sets (MAPSSWE p0.1) - On TS2, MFA FE significantly better only on test
sets 5 and 14
13EXPERIMENTAL RESULTS
MICROPHONE VARIATION
- Train on Sennheiser mic. evaluate on secondary
mic. - Matched conditions result in optimal performance
- Significant degradation for all front ends on
mismatched conditions - Both QIO and MFA provide improved robustness
relative to MFCC baseline
14EXPERIMENTAL RESULTS
ADDITIVE NOISE
15SUMMARY AND CONCLUSIONS
WHAT HAVE WE LEARNED?
- Front end specific parameter tuning did not
result in significant change in overall
performance (MFA still outperforms QIO) - Both QIO and MFA front ends handle convolution
and additive noise better than ETSI baseline - Both QIO and MFA front ends achieved ALV
evaluation goal of improving performance by at
least 25 relative over ETSI baseline - WER is still high ( 35), further research on
noise robust front end is needed
16SUMMARY AND CONCLUSIONS
AVAILABLE RESOURCES
17SUMMARY AND CONCLUSIONS
BRIEF BIBLIOGRAPHY
- N. Parihar, Performance Analysis of Advanced
Front Ends, M.S. Dissertation, Mississippi State
University, December 2003. - N. Parihar, J. Picone, D. Pearce, and H.G.
Hirsch, Performance Analysis of the Aurora Large
Vocabulary Baseline System, submitted to the
Eurospeech 2003, Geneva, Switzerland, September
2003. - N. Parihar and J. Picone, DSR Front End LVCSR
Evaluation - AU/384/02, Aurora Working Group,
European Telecommunications Standards Institute,
December 06, 2002. - D. Pearce, Overview of Evaluation Criteria for
Advanced Distributed Speech Recognition, ETSI
STQ-Aurora DSR Working Group, October 2001. - G. Hirsch, Experimental Framework for the
Performance Evaluation of Speech Recognition
Front-ends in a Large Vocabulary Task, ETSI
STQ-Aurora DSR Working Group, December 2002. - ETSI ES 201 108 v1.1.2 Distributed Speech
Recognition Front-end Feature Extraction
Algorithm Compression Algorithm, ETSI, April
2000.
18SUMMARY AND CONCLUSIONS
BIOGRAPHY
- Naveen Parihar is a M.S. student in Electrical
Engineering in the Department of Electrical and
Computer Engineering at Mississippi State
University. He currently leads the Core Speech
Technology team developing a state-of-the-art
public-domain speech recognition system. Mr.
Parihars research interests lie in the
development of discriminative algorithms for
better acoustic modeling and feature extraction.
Mr. Parihar is a student member of the IEEE. - Joseph Picone is currently a Professor in the
Department of Electrical and Computer Engineering
at Mississippi State University, where he also
directs the Institute for Signal and Information
Processing. For the past 15 years he has been
promoting open source speech technology. He has
previously been employed by Texas Instruments and
ATT Bell Laboratories. Dr. Picone received his
Ph.D. in Electrical Engineering from Illinois
Institute of Technology in 1983. He is a Senior
Member of the IEEE and a registered Professional
Engineer.