In speech evaluation, an Automatic Speech Recognition (ASR) model often computes time boundaries and phoneme posteriors for input features. However, limited data for ASR training hinders expansion of speech evaluation to low-resource languages. Open-source weakly-supervised models are capable of ASR over many languages, but they are frame-asynchronous and not phonemic, hindering feature extraction for speech evaluation. This paper proposes to overcome incompatibilities for feature extraction with weakly-supervised models, easing expansion of speech evaluation to low-resource languages. Phoneme posteriors are computed by mapping ASR hypotheses to a phoneme confusion network. Word instead of phoneme-level speaking rate and duration are used. Phoneme and frame-level features are combined using a cross-attention architecture, obviating phoneme time alignment. This performs comparably with standard frame-synchronous features on English speechocean762 and low-resource Tamil datasets.
Figures & tables
Fig. 1 : Transformer models for sentence-level speech evaluation
Number of
Pronunciation
PCC ↑ / MSE ↓ for Whisper feature
pronunciations
prior
CN-GOP
CN-GOP margin
LPP & LPR
1
no
0.24 / 0.95
0.28 / 0.99
0.42 / 0.16
up to 6
no
0.29 / 0.88
0.41 / 0.99
0.48 / 0.15
up to 6
yes
0.27 / 0.92
0.34 / 1.00
0.48 / 0.14
TABLE I : Phoneme pronunciation accuracy of per-phoneme SVRs trained on GOP, LPP, and LPR from Whisper, using multiple pronunciations and priors
Features
N-best
PCC / MSE for feature
from
size
GOP
GOP margin
LPP & LPR
hybrid
-
0.28 / 0.60
0.25 / 0.69
0.47 / 0.16
Whisper
10
0.31 / 0.91
0.37 / 0.99
0.49 / 0.17
100
0.30 / 0.90
0.41 / 1.00
0.47 / 0.15
200
0.29 / 0.88
0.41 / 0.99
0.48 / 0.15
TABLE II : Compare features from hybrid model and Whisper with different beam sizes, to compute phoneme pronunciation accuracy, using per-phoneme SVRs
Features from
GOP type
Use LPP & LPR
PCC
MSE
-
-
no
0.709
1.138
hybrid
GOP margin
yes
0.753
1.051
Whisper
CN-GOP margin
yes
0.748
1.042
Whisper
CN-GOP
yes
0.745
1.026
TABLE III : Sentence-level pronunciation accuracy of phoneme pooling model on GOP, LPP, and LPR features from Whisper
Use SR and ND
PCC
MSE
no
0.729
1.125
hybrid phoneme level
0.753
1.051
Whisper word level
0.743
1.031
TABLE IV : Inclusion of SR and ND features from Whisper
Features from
Combine features by
PCC
MSE
hybrid
phoneme pooling
0.753
1.051
cross-attention
0.718
1.095
Whisper
cross-attention
0.735
1.051
+ restrict attention to word
0.747
1.027
TABLE V : Combine per-phoneme and per-frame features either by averaging over the frames for each phoneme or by using cross-attention
Use word-level GOP
PCC
MSE
no
0.747
1.027
CN-WGOP margin
0.745
1.040
CN-WGOP
0.737
1.077
TABLE VI : Inclusion of word GOP features in cross-attention model
Features from
Combine features by
PCC
MSE
hybrid
phoneme pooling
0.547
0.637
cross-attention
0.611
0.699
Whisper
cross-attention
0.611
0.573
+ restrict attention to word
0.629
0.569
TABLE VII : Apply Whisper features and cross-attention model to low-resource Tamil language
N-best size
1-best WER (%) ↓
CN decode WER (%)
1 (greedy)
17.40
-
10
16.25
15.66
100
16.18
15.41
200
16.22
15.36
TABLE VIII : Confusion network decoding of Whisper N-best list
We present a phoneme-level analysis of automatic speech recognition (ASR) for two low-resourced and phonologically complex East Caucasian languages, Archi and Rutul, based on curated and standardized speech-transcript resources totaling approximately 50 minutes and 1 hour 20 minutes of audio, respectively. Existing recordings and transcriptions are consolidated and processed into a form suitable for ASR training and evaluation. We evaluate several state-of-the-art audio and audio-language models, including wav2vec2, Whisper, and Qwen2-Audio. For wav2vec2, we introduce a language-specific phoneme vocabulary with heuristic output-layer initialization, which yields consistent improvements and achieves performance comparable to or exceeding Whisper in these extremely low-resource settings. Beyond standard word and character error rates, we conduct a detailed phoneme-level error analysis. We find that phoneme recognition accuracy strongly correlates with training frequency, exhibiting a characteristic sigmoid-shaped learning curve. For Archi, this relationship partially breaks for Whisper, pointing to model-specific generalization effects beyond what is predicted by training frequency. Overall, our results indicate that many errors attributed to phonological complexity are better explained by data scarcity. These findings demonstrate the value of phoneme-level evaluation for understanding ASR behavior in low-resource, typologically complex languages.
V. S. D. S. Mahesh Akavarapu, Michael Daniel, Gerhard Jäger
Forced alignment evaluation typically requires manually annotated timestamps, limiting large-scale and multilingual analysis. We introduce two corpus-level metrics based on self-supervised (SSL) speech representations for reference-free forced alignment evaluation: Phoneme-Cluster Mutual Information (PCMI) and Word Acoustic Consistency Score (WACS). PCMI measures agreement between aligned phoneme labels and clusters induced from SSL-speech representations, while WACS measures consistency of repeated word realizations using dynamic time warping similarity between word representation sequences. Using both random and systematic perturbations, we show that PCMI and WACS degrade consistently under alignment perturbations. We further analyze the metrics across multiple alignment systems on 85 languages from FLEURS, validate them against manually annotated alignments from 45 languages in DoReCo, and evaluate them on two phonologically complex low-resource languages. The metrics effectively separate high- and low-quality alignments and correlate strongly with timestamp-based alignment quality measures. Our results demonstrate that SSL-speech representations enable scalable, reference-free forced alignment evaluation. The metrics are available as an open-source Python package at https://github.com/mahesh-ak/forced-aligner-metrics.
V. S. D. S. Mahesh Akavarapu, Michael Daniel, Gerhard Jäger
Speech large language models (speech LLMs) perform well on automatic speech recognition (ASR) when sufficient paired speech-text data is available, but their performance degrades in low-resource settings. A cascaded pipeline that performs speech-to-phoneme (S2P) conversion followed by phoneme-to-grapheme (P2G) conversion has been shown to outperform end-to-end speech LLMs in this regime, suggesting that phoneme-mediated processing is beneficial when paired data is scarce. We propose phoneme-guided initialization, a simple method that uses this insight within an end-to-end framework: we pre-train the audio encoder on S2P and the LLM on P2G tasks, then connect them and fine-tune the full model end-to-end on the target ASR task. Experiments on Japanese (CSJ), Chinese (AISHELL-1), and two low-resource languages from Common Voice 25.0 (Tatar and Urdu) show that our method matches or outperforms both the cascaded S2P-P2G baseline and the end-to-end model without P2G initialization.
Ryo Magoshi, Shinsuke Sakai, Tatsuya Kawahara
Graduate School of Informatics, Kyoto University, Japan