REVIEW 5 major objections 5 minor 36 references
The paper claims that a 1.6–2.1M-parameter adapter between a frozen Whisper encoder and TinyLlama delivers better end-to-end ASR, NER, and sentiment analysis than heavier baselines, using LLM-generated labels to compensate for scarce data.
Reviewed by Pith at T0; open to challenge. T0 means a machine referee read the full paper against a public rubric. the ladder, T0–T4 →
A parameter-efficient adapter bridging Whisper and TinyLlama reports relative improvements in speech recognition, named entity recognition, and sentiment analysis on low-resource benchmarks.
T0 review reviewed 2026-08-05 challenge →
load-bearing objection Plausible lightweight adapter for speech+LLM SLU, but the headline numbers are undercut by inconsistent reporting and unmatched baselines. the 5 major comments →
SpeechLLM: Unified Speech and Language Model for Enhanced Multi-Task Understanding in Low Resource Settings
The pith
A machine-rendered reading of the paper's core claim, the machinery that carries it, and where it could break.
The reading
Core claim
The proposed SENSE-ASR (SpeechLLM) model is a two-module bridge: an adaptive average pooling layer that compresses Whisper's 1500×768 speech embeddings to 250 frames, layer normalization, and a linear projection to 2048 dimensions so TinyLlama can consume them as if they were token embeddings. The paper's central discovery claim is that this simple adapter, trained with a blended loss that includes a CNN classifier regularizer and with optional LoRA on the LLM, produces end-to-end speech understanding at or above the level of heavier baselines: 74.6 SLUE score versus 68.1 for the comparable E2E baseline, a 9.5% relative improvement, while closing the usual gap between end-to-end and pipeline
What carries the argument
The adapter: adaptive average pooling reduces Whisper's 1500×768 embeddings to 250 frames, layer normalization stabilizes them, and a linear layer projects them to TinyLlama's 2048-dimensional embedding space. A three-layer 2D CNN classifier reads the projected speech embeddings and adds a task-specific loss (weighted binary cross-entropy for NER, cross-entropy for SA) blended with the LLM's token-level cross-entropy via alpha, as a regularizer. The training pipeline is the other mechanism: pretrain adapter on LibriSpeech ASR, continue on 960 hours of synthetically NER-annotated LibriSpeech, then fine-tune on 15 hours of SLUE-VoxPopuli or 12.8 hours of SLUE-VoxCeleb.
Load-bearing premise
The reported gains are credited to the adapter even though no comparison pairs this adapter with the same speech encoder and language model as the baselines, so the adapter's specific contribution is not actually isolated.
What would settle it
Run the proposed adapter on the exact backbone of the strongest baseline (for example, Whisper encoder with Llama-2-7B, or the SLAM-ASR adapter with TinyLlama) and compare WER on LibriSpeech test-clean and F1 on the SLUE sets; if the 26% WER and 32% SA gains shrink or disappear, the adapter alone is not what produces them.
If this is right
- End-to-end ASR+NER+SA can be built from frozen public speech and language models plus a roughly 2M-parameter adapter, lowering training cost.
- LLM-generated synthetic NER labels for speech can substitute for large human-annotated datasets, reducing labeling expense.
- A CNN classifier used as a loss regularizer can improve downstream NER and SA F1 scores without adding inference cost.
- Light LoRA fine-tuning of the LLM yields further gains with only 0.58% of total parameters trained.
- E2E spoken language understanding can reach pipeline-level SLUE scores (74.6 vs 75.7 pipeline), narrowing the historical gap.
Where Pith is reading between the lines
- Because the SLAM-ASR baseline uses Llama-2-7B while the proposed model uses TinyLlama, and the SLUE baselines use wav2vec features with linear heads, the magnitude of the reported gains may be partly a function of backbone choice rather than the adapter architecture alone; a matched-backbone comparison would settle this.
- The synthetic annotation pipeline contributes both label noise and 64× more NER training data than the human-labeled set; ablating the amount of synthetic data (e.g., 100h vs 960h) would show whether gains saturate and where noise starts to hurt.
- The same frozen-encoder/small-adapter recipe is a plausible template for other low-resource spoken language tasks such as slot filling, emotion recognition, or language identification, since the adapter and regularizer are task-agnostic apart from the classifier head.
- The paper reports parameter counts but not training or inference time and memory; the practical value of the 7× reduction depends on those measurements in deployment.
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. The paper proposes SpeechLLM/SENSE-ASR, an end-to-end model that connects a frozen Whisper speech encoder to a frozen TinyLlama language model through a lightweight trainable adapter (adaptive average pooling + layer norm + linear projection), optionally with a convolutional classifier regularizer and LoRA fine-tuning. The method is evaluated on three tasks: LibriSpeech ASR, SLUE-VoxPopuli NER, and SLUE-VoxCeleb sentiment analysis. The authors also propose an LLM-based synthetic annotation pipeline for NER using GPT-4o to label 960 hours of LibriSpeech, with filtering and re-verification, and report large relative improvements over prior baselines (26% WER reduction on ASR, 6.3% NER F1 increase, 32% SA F1 increase) while using about 7x fewer trainable parameters.
Significance. If the results held, the contribution would be practically valuable: a parameter-efficient adapter enabling a frozen speech encoder and a small LLM to outperform larger, more heavily trained baselines on low-resource spoken language understanding would be a meaningful step. The synthetic NER annotation pipeline, with its explicit F1 validation on a human-annotated dev set, is a useful idea. However, the paper's central causal claim—that the adapter design, rather than the choice of backbone, training schedule, or decoding settings, is responsible for the gains—is not tested by the current experimental design. Internal numerical contradictions further reduce confidence. The reported numbers may still be correct, but the evidence as presented is not sufficient to support the abstract's strong causal and state-of-the-art claims.
major comments (5)
- [§3.1–§3.4, Tables 2–5] The central comparison is not controlled. In Table 2, SLAM-ASR pairs Whisper with Llama-2-7B, while SpeechLLM pairs Whisper with TinyLlama (~1.1B); no experiment holds the language model fixed and varies only the adapter. In Tables 3–5, the SLUE baselines use wav2vec 2.0 features with a linear head (plus an external TED LM), not an LLM decoder. The proposed system additionally uses a multi-stage training recipe (ASR pretraining, 960h synthetic NER pretraining, fine-tuning), different decoding parameters (beam 5, temperature 1.0, repetition penalty 2.0), and a different speech encoder. Any of these factors could explain the reported WER/F1 differences. To substantiate the claim that the proposed adapter is the source of the gains, the authors should run same-backbone ablations, e.g., replace the proposed adapter with a linear projection or a Q-Former-like adapter while keeping Whisper+Tin
- [§3.2, §3.3, Tables 3–5] Several numbers in the text contradict the tables. (a) §3.2 states the Whisper Medium 'Random' F1 is 61.5%, but Table 3 reports 62.0%. (b) §3.2 reports a LoRA F1 of 68.1% and Label F1 of 81.1%, while Table 3 reports 68.9% and 81.2%; the text later mentions 68.9% after beam decoding, but the table has only one LoRA row. (c) §3.2 reports a SLUE score of 79.2, while Table 5's best SLUE score is 74.6. (d) §3.3 says Whisper Small LS-ASR WER is 18.0%, but Tables 4 and 5 report 16.0%. (e) §3.3 says the LoRA SA configuration achieves WER 12.9%, but Table 4 reports 11.5%. These inconsistencies directly affect the paper's headline numbers and must be reconciled.
- [§3.4 and Conclusion] The conclusion states 'state-of-the-art F1 score of 74.6% on the SLUE benchmark.' The SLUE score is defined in §2.5 as an average of (100−WER) and F1 scores, not an F1 score; 74.6 is a SLUE score, not an F1 metric. Also, calling the SLUE 'E2E Approach' rows 'equivalent E2E benchmark model' is misleading: those rows use wav2vec 2.0 with a linear head and an external LM, not an LLM. The authors should rename and clarify the benchmark model class, and avoid the term 'equivalent.'
- [§3.1, Table 2] The parameter-efficiency claim is imprecise. '7x fewer trainable parameters' is roughly consistent with 15.2/2.1≈7.2, but the accompanying text says 'only 15% of the trainable parameters'; 2.1/15.2 is 13.8% and 1.6/13.1 is 12.2%. More importantly, the baseline parameter counts in Table 2 are presented without definition: are these trainable parameters of the adapter, or total trainable parameters of the whole model? Clarify so that the comparison is meaningful.
- [§3.1–§3.4] No error bars, confidence intervals, or repeated-run statistics are reported. Several claimed improvements are small (e.g., NER F1 65.0 vs 65.8; SA 63.6 vs 65.9) and could lie within run-to-run variance, especially for a stochastic training pipeline with multi-stage pretraining. The authors should either report multiple seeds or at least state the number of runs and variance for the main rows.
minor comments (5)
- [Title/§2] The paper title is 'SpeechLLM' but the model is called 'SENSE-ASR' throughout; the abstract and index terms also use 'SpeechLLM.' Unify the naming.
- [§2.5] 'Jon ̸= Jhon' appears to be a typo for 'John ≠ Jhon' (or 'Jon ≠ John'). Also the entity example uses inconsistent spellings.
- [§3.2] 'a5.69% relative improvement' is missing a space; should read 'a 5.69% relative improvement.'
- [Abstract] The abstract states 'SLUE score improvement of 6.6% and 9.5%' but the text in §3.4 reports 2% and 2.7% relative improvements for classifier and LoRA additions. Clarify which baseline each percentage refers to.
- [§3.4] The narrative says the best SLUE score is '74.3%, 2.7% relative improvement' then later '74.6'; these should be made consistent.
Circularity Check
No circularity: reported gains are measured on held-out LibriSpeech/SLUE-V test sets against external published baselines; no self-citations or fitted-to-test quantities.
full rationale
The paper's central results are empirical evaluations against external benchmarks, not quantities derived by construction from their inputs. The adapter is trained only on train splits (LibriSpeech 960h; SLUE-V fine-tune) and scored on LibriSpeech test-clean/test-other and SLUE-V test splits (Tables 2-5). The claimed improvements — 26% WER (Table 2: (5.01-3.67)/5.01), 6.3% NER F1 (Table 3: 64.8 to 68.9 vs the SLUE E2E benchmark), 32% SA F1 (Table 4: 49.8 to 65.9), 9.5% SLUE (68.1 to 74.6) — are arithmetic differences between two independently measured, published or held-out values. The synthetic NER pretraining data is a training input (GPT-4o annotations on LibriSpeech), and the prompt selection for it uses only the SLUE-V dev split, not the test split; the classifier regularizer uses training ground truth and is discarded at inference (Sec. 2.4.3). No self-citation is load-bearing: all 31 references are by other authors, and the adapter design (adaptive pooling + layer norm + linear projection) is presented as the paper's own contribution rather than imported from a prior-work theorem. This is therefore a 0: the derivation chain is self-contained against external test sets. Two non-circularity concerns belong to correctness risk, not circularity: (1) the causal claim that the adapter alone drives the gains is not directly tested, since the ASR baseline SLAM-ASR uses Llama-2-7B and the SLUE baselines use wav2vec2 + linear head rather than matched backbones; (2) internal numerical inconsistencies (Table 3 shows Random F1=62.0 where text says 61.5; Sec 3.2 reports SLUE 79.2 vs Table 5's 74.6; Sec 3.3 text reports Whisper-Small LS-ASR WER 18.0 vs Table 4's 16.0) reduce confidence in number stability but do not make any result equal-by-construction to its input.
Axiom & Free-Parameter Ledger
free parameters (6)
- Pooling size =
250
- Classifier regularizer weight alpha =
0.2
- LoRA rank and alpha =
r=32, alpha=32
- Inference decoding parameters =
beam=5, temp=1.0, rep_penalty=2.0, len_penalty=0.5
- NER synthetic labeling few-shot examples =
1000 balanced examples + 7 entity categories
- Training schedules =
epochs, LR, batch sizes, warmup (e.g., 20 epochs, 1e-4, batch 4, 1000 warmup)
axioms (5)
- domain assumption Whisper encoder outputs contain enough semantic information for NER/sentiment after linear projection into LLM space
- domain assumption TinyLlama can jointly process concatenated speech tokens, prompts, and ground-truth tokens and perform ASR, NER, and SA generation
- domain assumption GPT-4o few-shot annotation of LibriSpeech transcripts provides labels close enough to human NER labels to pretrain the adapter
- domain assumption Adaptive average pooling to 250 tokens preserves the information needed for NER and sentiment
- domain assumption The classifier regularizer (BCE for NER, CE for SA) with alpha=0.2 does not degrade generation quality
Cite this review
Pith. "Pith review of SpeechLLM: Unified Speech and Language Model for Enhanced Multi-Task Understanding in Low Resource Settings." pith.science (2026). https://pith.science/paper/FAJGYN2F
@misc{pith2026250904473,
author = {Pith},
title = {Pith review of: SpeechLLM: Unified Speech and Language Model for Enhanced Multi-Task Understanding in Low Resource Settings},
year = {2026},
howpublished = {\url{https://pith.science/paper/FAJGYN2F}},
note = {Machine review of arXiv:2509.04473}
}
read the original abstract
While integrating speech encoder with LLM requires substantial data and resources, use cases face limitations due to insufficient availability. To address this, we propose a solution with a parameter-efficient adapter that converts speech embeddings into LLM-compatible tokens, focusing on end-to-end automatic speech recognition (ASR), named entity recognition (NER), and sentiment analysis (SA). To reduce labeling costs, we employ an LLM-based synthetic dataset annotation technique. The proposed adapter, using 7x fewer trainable parameters, achieves significant performance gains: a 26% relative Word Error Rates (WER) improvement on the LibriSpeech ASR task, a 6.3% relative F1 score increase on the NER task, and a 32% relative F1 score boost on the SA task. Moreover, using advanced techniques such as adding a classifier regularizer and optimizing the LLM with Low-Rank Adaptation (LoRA) yields notable performance gains, with Spoken Language Understanding Evaluation (SLUE) score improvement of 6.6% and 9.5%
Figures
Reference graph
Works this paper leans on
-
[1]
Introduction Recent advancements in speech and language tasks have shown a significant shift in interest from domain-specific models to those trained on much larger and more general corpora. In par- ticular, Large Language Models (LLMs) have demonstrated a remarkable ability to perform a wide variety of tasks and show great promise for more general langua...
work page internal anchor Pith review Pith/arXiv arXiv 2025
-
[2]
Model Architecture 2.1. Overview of SpeechLLM The proposed SENSE-ASR architecture, as shown in Fig.1, in- tegrates a frozen Whisper speech encoder with a pre-trained TinyLlama language model using a simple trainable adapter. The training input to the SENSE-ASR consists of three com- ponents: speech, task instruction prompt, and the ground truth text. Give...
work page 2048
-
[3]
Experimental Results and Discussion In this section we evaluate the proposed SENSE-ASR model using the Librispeech, SLUE-V oxPopuli(VP) and SLUE- V oxCeleb(VC) dataset against the baselines presented in [5, 23] using metrics discussed in 2.5. The ASR baseline benchmarks on the Librispeech dataset are derived from the study in [5], which is similar to ours...
work page 2000
-
[4]
We utilized 12.8 hours of training data and 3.2 hours of de- velopment data for training and tuning tasks, and assessed the model’s performance on a 7.8-hour test split, as reported in this paper. We conduct the SA training for 50 epochs with a learning rate 5 ∗ 10−4, batch size 6, and a linear decay scheduler with 3000 warm-up steps. Speech Model Pretrai...
-
[5]
Conclusion In this paper, we introduced SENSE-ASR, an E2E model that effectively integrates speech and large language models using a novel adapter design to enhance performance on ASR, NER and SA tasks. The proposed model exhibits the capability to capture semantic meanings by effectively mapping speech features to text tokens that are interpretable by LL...
-
[6]
J. Achiam, S. Adler, S. Agarwal, L. Ahmad, I. Akkaya, F. L. Ale- man, D. Almeida, J. Altenschmidt, S. Altman, S. Anadkat et al., “Gpt-4 technical report,” arXiv preprint arXiv:2303.08774, 2023
Pith/arXiv arXiv 2023
-
[7]
A survey on speech large language models,
J. Peng, Y . Wang, Y . Fang, Y . Xi, X. Li, X. Zhang, and K. Yu, “A survey on speech large language models,” 2025. [Online]. Available: https://arxiv.org/abs/2410.18908
arXiv 2025
-
[8]
Robust speech recognition via large-scale weak supervision,
A. Radford, J. W. Kim, T. Xu, G. Brockman, C. McLeavey, and I. Sutskever, “Robust speech recognition via large-scale weak supervision,” in International conference on machine learning . PMLR, 2023, pp. 28 492–28 518
2023
-
[9]
Tinyllama: An open- source small language model,
P. Zhang, G. Zeng, T. Wang, and W. Lu, “Tinyllama: An open- source small language model,” arXiv preprint arXiv:2401.02385, 2024
Pith/arXiv arXiv 2024
-
[10]
An embarrassingly simple approach for llm with strong asr capacity,
Z. Ma, G. Yang, Y . Yang, Z. Gao, J. Wang, Z. Du, F. Yu, Q. Chen, S. Zheng, S. Zhanget al., “An embarrassingly simple approach for llm with strong asr capacity,” arXiv preprint arXiv:2402.08846 , 2024
Pith/arXiv arXiv 2024
-
[11]
Speechgpt: Empowering large language models with intrinsic cross-modal conversational abilities,
D. Zhang, S. Li, X. Zhang, J. Zhan, P. Wang, Y . Zhou, and X. Qiu, “Speechgpt: Empowering large language models with intrinsic cross-modal conversational abilities,” 2023. [Online]. Available: https://arxiv.org/abs/2305.11000
Pith/arXiv arXiv 2023
-
[12]
Salmonn: Towards generic hearing abilities for large language models,
C. Tang, W. Yu, G. Sun, X. Chen, T. Tan, W. Li, L. Lu, Z. Ma, and C. Zhang, “Salmonn: Towards generic hearing abilities for large language models,” arXiv preprint arXiv:2310.13289, 2023
Pith/arXiv arXiv 2023
-
[13]
Qwen-audio: Advancing universal audio understand- ing via unified large-scale audio-language models,
Y . Chu, J. Xu, X. Zhou, Q. Yang, S. Zhang, Z. Yan, C. Zhou, and J. Zhou, “Qwen-audio: Advancing universal audio understand- ing via unified large-scale audio-language models,”arXiv preprint arXiv:2311.07919, 2023
Pith/arXiv arXiv 2023
-
[14]
Anygpt: Unified multi- modal llm with discrete sequence modeling,
J. Zhan, J. Dai, J. Ye, Y . Zhou, D. Zhang, Z. Liu, X. Zhang, R. Yuan, G. Zhang, L. Li et al. , “Anygpt: Unified multi- modal llm with discrete sequence modeling,” arXiv preprint arXiv:2402.12226, 2024
Pith/arXiv arXiv 2024
-
[15]
J. Yu, X. Geng, Y . Li, M. Ren, W. Tang, J. Li, Z. Lan, M. Zhang, H. Yang, S. Huang et al., “” i’ve heard of you!”: Generate spo- ken named entity recognition data for unseen entities,” inICASSP 2025-2025 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2025, pp. 1–5
work page 2025
-
[16]
Q. Meeus, M.-F. Moens, and H. Van Hamme, “Whisper-slu: Ex- tending a pretrained speech-to-text transformer for low resource spoken language understanding,” in2023 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU). IEEE, 2023, pp. 1–6
work page 2023
-
[17]
On the eval- uation of speech foundation models for spoken language under- standing,
S. Arora, A. Pasad, C.-M. Chien, J. Han, R. Sharma, J.-w. Jung, H. Dhamyal, W. Chen, S. Shon, H.-y. Lee et al., “On the eval- uation of speech foundation models for spoken language under- standing,” arXiv preprint arXiv:2406.10083, 2024
Pith/arXiv arXiv 2024
-
[18]
S. Arora, H. Futami, J.-w. Jung, Y . Peng, R. Sharma, Y . Kashi- wagi, E. Tsunoo, K. Livescu, and S. Watanabe, “Universlu: Uni- versal spoken language understanding for diverse tasks with nat- ural language instructions,” in Proceedings of the 2024 Confer- ence of the North American Chapter of the Association for Com- putational Linguistics: Human Languag...
work page 2024
-
[19]
Prompting Whisper for QA-driven Zero-shot End-to-end Spoken Language Understanding
M. Li, S. Keizer, and R. Doddipatla, “Prompting whisper for qa-driven zero-shot end-to-end spoken language understanding,” arXiv preprint arXiv:2406.15209, 2024
work page internal anchor Pith review Pith/arXiv arXiv 2024
-
[20]
Z. Chen, H. Huang, A. Andrusenko, O. Hrinchuk, K. C. Puvvada, J. Li, S. Ghosh, J. Balam, and B. Ginsburg, “Salm: Speech- augmented language model with in-context learning for speech recognition and translation,” in ICASSP 2024-2024 IEEE Inter- national Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2024, pp. 13 521–13 525
work page 2024
-
[21]
WhisperNER: Unified Open Named Entity and Speech Recognition
G. Ayache, M. Pirchi, A. Navon, A. Shamsian, G. Hetz, and J. Keshet, “Whisperner: Unified open named entity and speech recognition,” arXiv preprint arXiv:2409.08107, 2024
work page internal anchor Pith review Pith/arXiv arXiv 2024
-
[22]
Chinese asr and ner improvement based on whisper fine-tuning,
H. Yang, M. Zhang, S. Tao, M. Ma, and Y . Qin, “Chinese asr and ner improvement based on whisper fine-tuning,” in 2023 25th In- ternational Conference on Advanced Communication Technology (ICACT). IEEE, 2023, pp. 213–217
work page 2023
-
[23]
Nuner: Entity recognition encoder pre-training via llm-annotated data,
S. Bogdanov, A. Constantin, T. Bernard, B. Crabb ´e, and E. Bernard, “Nuner: Entity recognition encoder pre-training via llm-annotated data,” 2024. [Online]. Available: https: //arxiv.org/abs/2402.15343
Pith/arXiv arXiv 2024
-
[24]
Using Large Language Model for End-to-End Chinese ASR and NER
Y . Li, J. Yu, M. Zhang, M. Ren, Y . Zhao, X. Zhao, S. Tao, J. Su, and H. Yang, “Using large language model for end-to-end chinese asr and ner,” arXiv preprint arXiv:2401.11382, 2024
work page internal anchor Pith review Pith/arXiv arXiv 2024
-
[25]
Wavllm: Towards ro- bust and adaptive speech large language model,
S. Hu, L. Zhou, S. Liu, S. Chen, L. Meng, H. Hao, J. Pan, X. Liu, J. Li, S. Sivasankaran et al. , “Wavllm: Towards ro- bust and adaptive speech large language model,” arXiv preprint arXiv:2404.00656, 2024
Pith/arXiv arXiv 2024
-
[26]
End-to-end named entity recognition from english speech,
H. Yadav, S. Ghosh, Y . Yu, and R. R. Shah, “End-to-end named entity recognition from english speech,” 2020. [Online]. Available: https://arxiv.org/abs/2005.11184
Pith/arXiv arXiv 2020
-
[27]
End-to-end named entity and semantic concept extraction from speech,
S. Ghannay, A. Caubri `ere, Y . Est`eve, N. Camelin, E. Simonnet, A. Laurent, and E. Morin, “End-to-end named entity and semantic concept extraction from speech,” in 2018 IEEE Spoken Language Technology Workshop (SLT). IEEE, 2018, pp. 692–699
work page 2018
-
[28]
Slue: New benchmark tasks for spoken language un- derstanding evaluation on natural speech,
S. Shon, A. Pasad, F. Wu, P. Brusco, Y . Artzi, K. Livescu, and K. J. Han, “Slue: New benchmark tasks for spoken language un- derstanding evaluation on natural speech,” in ICASSP 2022-2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2022, pp. 7927–7931
work page 2022
-
[29]
Lib- rispeech: an asr corpus based on public domain audio books,
V . Panayotov, G. Chen, D. Povey, and S. Khudanpur, “Lib- rispeech: an asr corpus based on public domain audio books,” in 2015 IEEE international conference on acoustics, speech and signal processing (ICASSP). IEEE, 2015, pp. 5206–5210
work page 2015
-
[30]
C. Wang, M. Riviere, A. Lee, A. Wu, C. Talnikar, D. Haz- iza, M. Williamson, J. Pino, and E. Dupoux, “V oxpopuli: A large-scale multilingual speech corpus for representation learn- ing, semi-supervised learning and interpretation,” arXiv preprint arXiv:2101.00390, 2021
Pith/arXiv arXiv 2021
-
[31]
V oxceleb: a large-scale speaker identification dataset,
A. Nagrani, J. S. Chung, and A. Zisserman, “V oxceleb: a large-scale speaker identification dataset,” arXiv preprint arXiv:1706.08612, 2017
Pith/arXiv arXiv 2017
-
[32]
E. Hovy, M. Marcus, M. Palmer, L. Ramshaw, and R. Weischedel, “Ontonotes: the 90% solution,” in Proceedings of the human lan- guage technology conference of the NAACL, Companion Volume: Short Papers, 2006, pp. 57–60
work page 2006
-
[33]
Promptner: Prompting for named entity recognition,
D. Ashok and Z. C. Lipton, “Promptner: Prompting for named entity recognition,” arXiv preprint arXiv:2305.15444, 2023
Pith/arXiv arXiv 2023
-
[34]
Specaugment: A simple data augmentation method for automatic speech recognition,
D. S. Park, W. Chan, Y . Zhang, C.-C. Chiu, B. Zoph, E. D. Cubuk, and Q. V . Le, “Specaugment: A simple data augmentation method for automatic speech recognition,” in Interspeech 2019 . ISCA, Sep. 2019. [Online]. Available: http://dx.doi.org/10.21437/Interspeech.2019-2680
-
[35]
wav2vec 2.0: A framework for self-supervised learning of speech repre- sentations,
A. Baevski, Y . Zhou, A. Mohamed, and M. Auli, “wav2vec 2.0: A framework for self-supervised learning of speech repre- sentations,” Advances in neural information processing systems , vol. 33, pp. 12 449–12 460, 2020
2020
-
[36]
Hubert: Self-supervised speech represen- tation learning by masked prediction of hidden units,
W.-N. Hsu, B. Bolte, Y .-H. H. Tsai, K. Lakhotia, R. Salakhutdi- nov, and A. Mohamed, “Hubert: Self-supervised speech represen- tation learning by masked prediction of hidden units,”IEEE/ACM transactions on audio, speech, and language processing, vol. 29, pp. 3451–3460, 2021
2021
This paper was first reviewed by deepseek-v4-flash on August 5, 2026.
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