Jul 2026· PHM Society European Conference· Vol 9, pp. 1-12· 0 citations· 30 references
TL;DR
Results show consistent gains from lightweight domain adaptation on both held-out synthetic data and real-world recordings, confirming that synthetic data generation combined with LoRA-based fine-tuning is an effective and computationally practical strategy for improving ASR accuracy in specialized technical domains where labeled speech is scarce.
Abstract
Automatic speech recognition (ASR), or speech-to-text (STT), is becoming an important interface for AI systems in diagnostic workflows, but general-purpose ASR models often degrade in specialized technical domains. In diagnostic applications such as fault identification, root cause analysis, and repair recommendation, general-purpose ASR systems struggle with domain-specific terminology, abbreviations, part identifiers, and measurement expressions, leading to elevated transcription errors. This work presents a domain adaptation pipeline that unifies three components: a synthetic benchmarking framework in which domain-specific technical text is converted to speech via text-to-speech~(TTS) synthesis and transcribed by open-source ASR models to establish baseline performance; Low-Rank Adaptation~(LoRA)-based fine-tuning of Whisper Large-v3 using those synthetic audio-text pairs; and transfer validation on curated real-world automotive YouTube recordings to assess generalization beyond synthetic conditions. Using automotive technical language as a representative diagnostic domain, a data-scaling study employing progressively larger subsets of in-domain training data evaluates performance on a held-out test set via word error rate~(WER), character error rate~(CER), normalized error metrics, alphanumeric error rate, semantic similarity, and Bidirectional Encoder Representations from Transformers Score~(BERTScore). Results show consistent gains from lightweight domain adaptation on both held-out synthetic data and real-world recordings, confirming that synthetic data generation combined with LoRA-based fine-tuning is an effective and computationally practical strategy for improving ASR accuracy in specialized technical domains where labeled speech is scarce.
End-to-end on-device Automatic Speech Recognition (ASR) systems have demonstrated remarkable accuracy and efficiency in recent years. However, challenges persist in correctly transcribing infrequent named entities (e.g., geographical locations, business entities, person names, etc.) and handling diverse user accents, which remain underrepresented in training datasets. While information retrieval augmentation or Retrieval Augmentation Generation (RAG) for correction of named entities has shown promise in knowledge-grounded NLP tasks when paired with large language models (LLMs), its integration into real-time on-device systems is non-trivial due to computational constraints. We introduce a novel lightweight method combining phonetic-aware retrieval, vector-based semantic search and generative correction. The system leverages a lightweight phonetic index for rapid candidate entity retrieval and a dense vectorembedding module to refine predictions as well as model the ASR error output distribution in generative space. Additionally, we introduce a novel approach to model ASR errors in natural language. Experiments on test sets emphasizing place names, monuments, airports, and landscapes yielded an increase in correct Named Entity (NE) recognition accuracy by 9.6% compared to baseline. These gains underscore the efficacy of hybrid retrieval-generation paradigms in resource-constrained environments.
Kiranmayi Gandikota, Anunay Katare, C. Pandey et al.· International Conference on...· 0 citations
Synthetic speech provides scalable supervision for automatic speech recognition (ASR), but its benefit depends on the selected texts, reference speech, and amount of synthesized data. We present a unified phoneme-based TTS-to-ASR augmentation pipeline built around a multilingual TTS model trained from scratch using the F5-TTS architecture with language-ID conditioning. The pipeline combines language-specific grapheme-to-phoneme conversion, reference-speech filtering, candidate-text selection, synthesis, and matched ASR continuation. We further propose phoneme-frequency-guided selection (PFGS), which ranks candidate sentences using phoneme frequencies estimated from real ASR training labels. Experiments with separate monolingual ASR systems for Arabic, French, Italian, and Portuguese span 13 test sets. Across the synthesis-scale sweep, random augmentation improves over matched real-only continuation on 11 test sets. Under a nominal 60% synthesis budget, PFGS improves over real-only training on 12 test sets and over random selection on 9. Its largest relative word error rate (WER) reduction against random selection is 19.3%. With target texts and synthesis counts fixed, reference-speech filtering reduces absolute WER by 0.29 and 0.59 points on Italian and French Common Voice, respectively. These results identify synthesis scale, candidate-text content, and reference quality as important control variables in TTS-based ASR augmentation.
Zhen Wang, Tian-Rui Wu, Rong-Qi Han et al.· 0 citations
Experimental results show that VALL-E outperforms the state-of-the-art zero-shot TTS system in terms of speech naturalness and speaker similarity and could preserve the speaker’s emotion and acoustic environment from the prompt in synthesis.
Few-step diffusion and flow-matching text-to-speech (TTS) models are usually trained with local objectives, such as conditional flow matching, reconstruction, and stop prediction. These losses provide stable optimization, but they never ask whether sampled speech follows the distribution of high-quality speech. We propose Speech Representation Fr'echet Distance loss (SR-FD), a training-time distributional regularizer for tokenizer-free flow-matching autoregressive TTS. During fine-tuning, the model synthesizes speech with the same few-step sampler used at deployment, and SR-FD matches the mean and covariance of frozen Whisper and CTC features of this speech to reference statistics computed offline from three complementary content targets. The loss requires no discriminator and no inference-time computation. On Seed-TTS English, four-step SR-FD fine-tuning reduces WER from the original four-step VoxCPM2 baseline's 2.2279% to 1.4147%, a 36.5% relative reduction, and also surpasses the original ten-step baseline at 1.7366%; both gains are significant under an utterance-level paired bootstrap. Speaker similarity and objective quality proxies are preserved at the ten-step level, and an error analysis shows the gain comes from content substitutions across all prompt lengths. SR-FD is thus an intelligibility-improving distributional regularizer for few-step TTS.
Ho-Lam Chung, Kuan-Po Huang, Bo-Ru Lu et al.· 0 citations
Audio-encoder-LLM-decoder architectures have become the dominant paradigm for modern automatic speech recognition (ASR), improving transcription quality through large-scale language modeling. However, the cost of autoregressive decoding scales with decoder size, creating a fundamental trade-off between recognition quality and serving latency. We argue this trade-off is not inherent: unlike open-ended text generation, ASR outputs are strongly anchored to the input speech signal, providing a natural inductive bias toward high-parallelism decoding. Building on this, we introduce ParaASR, an ASR system that leverages Multi-Token Prediction (MTP) to let a 4B LLM decoder emit multiple tokens per forward step. Starting from a publicly available audio-language foundation, the model first establishes a robust autoregressive recognizer and then aligns five future-token branches through a staged optimization recipe. At inference, it proposes a six-token continuation per step and admits only the verified prefix into the transcript, preserving the safety of standard autoregressive decoding. The average accepted length reaches 5.0 out of 6 proposed tokens, confirming that the deterministic structure of speech makes ASR an especially natural setting for multi-token decoding. ParaASR further retains a native 32K-context window and transcribes up to 30 minutes of audio in a single pass. Across diverse benchmarks, it attains average error rates of 2.97%, 3.68%, and 3.70% on Chinese, English, and long-form evaluations, respectively, while reaching a real-time factor (RTF) as low as 0.0053. These results show that decoder scaling, low-latency inference, and long-context transcription need not be competing goals when future-token proposals are anchored by the acoustic signal and guarded by autoregressive verification.
Qingjian Lin, Yuxin Li, Haoyang Zhang et al.· 1 citation
Automatic speech recognition (ASR) systems have achieved high accuracy with transformer-based models, enabling deployment in critical applications. However, they remain vulnerable to adversarial manipulation, particularly in black-box settings where attacks must preserve perceptual naturalness. This work introduces GATAS, a black-box testing approach that generates failure inducing inputs by operating in the phoneme-level latent space of a text- to-speech model. Instead of perturbing waveforms directly, the approach interpolates latent representations to induce transcription errors while remaining within the manifold of natural speech. The attack is formulated as a multi-objective optimization problem balancing semantic divergence and perceptual quality. Our empirical evaluation against both white-box and black-box baselines shows that GATAS achieves a 98% success rate while producing lower distortion and higher perceptual quality, as confirmed by human studies. Despite operating without gradient access, GATAS remains competitive against white-box methods, highlighting that representation and perceptual alignment are more critical than access to model internals. Overall, our results demonstrate that untargeted latent-space optimization enables the efficient generation of realistic and effective test cases for ASR systems.
Yanis Xabier Wilbrand Pena, Oliver Weissl, Andrea Stocco· 0 citations