Large Audio-Language Models (LALMs) typically struggle with localized dialectal prosody due to the scarcity of specialized corpora. We present TW-Sound580K, a Taiwanese audio-text instruction dataset developed through a Verify-Generate-Critique (VGC) protocol. This pipeline leverages Dual-ASR validation to filter 522K raw clips, subsequently expanding them into 580,000 high-fidelity instruction pairs using a teacher model. The dataset's utility is demonstrated through Tai-LALM, which fine-tunes a DeSTA 2.5-Audio-initialized backbone and incorporates a dynamic Dual-ASR Arbitration strategy to optimize transcription selection during inference. On the TAU Benchmark, Tai-LALM reaches 49.1% accuracy, marking a 6.5% absolute improvement over the zero-shot baseline (42.6% with ASR text conditioning). This confirms that integrating regional corpora with rigorous curation and dynamic arbitration significantly enhances LALM performance on localized speech.
Figures & tables
Figure 1: The proposed framework for TW-Sound580K dataset construction and Tai-LALM fine-tuning, illustrating the DeSTA 2.5-Audio-based localization pipeline.
Figure 2: Label occurrence distribution in the TW-Sound580K dataset.
System
Params
Single
Multi
Overall
Gemini 2.5 Pro (Teacher)
–
72.4
73.9
73.0
Gemini 2.5 Flash
–
61.3
63.2
62.1
Qwen2.5-Omni-7B [ 32 ]
7.6B
46.4
46.1
46.3
DeSTA 2.5-Audio [ 24 ]
8.8B
43.3
41.7
42.6
Qwen2-Audio-Instruct [ 33 ]
8.2B
30.3
27.8
29.3
Gemma-3n-E2B-it
4.4B
29.6
25.8
28.0
Table 1: Performance on the TAU Benchmark. Tai-LALM significantly outperforms zero-shot baselines and the naive SFT control.
Figure 3: Scaling law analysis demonstrating the efficacy of our localized data pipeline on the TW-Sound580K dataset.
Strategy
SFT Data
ASR Injection
TAU Acc (%)
Zero-shot Baseline
—
None
41.1
Direct SFT (Baseline)
Unfiltered Raw TW-580K+
None
43.8
Single-ASR
Unfiltered Raw TW-580K+
SenseVoice
44.5
Single-ASR(Negative Control)
Unfiltered Raw TW-580K+
Whisper-v3
46.4
Dual-ASR
Unfiltered Raw TW-580K+
Dual-ASR (AC-PPL)
47.5
Ours (VGC Pipeline)
Filtered TW-580K
Dual-ASR (AC-PPL)
49.1
Table 2: Ablation study evaluating the impact of data filtering and inference-time ASR injection strategies on TAU Benchmark accuracy.
Model
LibriSpeech
ESC-50
CREMA-D
(WER% ↓ )
(Acc% ↑ )
(Acc% ↑ )
DeSTA 2.5-Audio (Zero-shot)
4.71
34.50
75.22
Tai-LALM (Ours)
3.92
33.50
72.28
Table 3: General audio evaluation. Tai-LALM improves LibriSpeech ASR and maintains competitive non-speech performance, confirming no catastrophic forgetting after localized fine-tuning.
Taiwanese Hokkien (Taigi) presents unique opportunities for advancing speech technology methodologies that can generalize to diverse linguistic contexts. We introduce Breeze Taigi, a comprehensive framework centered on standardized benchmarks for evaluating Taigi speech recognition and synthesis systems. Our primary contribution is a reproducible evaluation methodology that leverages parallel Taiwanese Mandarin resources. We provide 30 carefully curated Mandarin-Taigi audio pairs from Taiwan's Executive Yuan public service announcements with normalized ground truth transcriptions. We establish Character Error Rate (CER) as the standard metric and implement normalization procedures to enable fair cross-system comparisons. To demonstrate the benchmark's utility and provide reference implementations, we develop speech recognition and synthesis models through a methodology that leverages existing Taiwanese Mandarin resources and large-scale synthetic data generation. In particular, we fine-tune a Whisper model on approximately 10,000 hours of Taigi synthetic speech data. Our ASR model achieves 30.13% average CER on the benchmark, outperforming existing commercial and research systems. By providing standardized evaluation protocols, diverse training datasets, and open baseline models, we offer a replicable framework with methodologies applicable to various linguistic contexts.
Yu-Siang Lan, Chia-Sheng Liu, Yi-Chang Chen +5
National Yang Ming Chiao Tung University · Work done at MediaTek Research · MediaTek Research
Large Audio-Language Models (LALMs) integrate audio perception and language understanding within a unified framework, enabling a wide range of real-world applications. Despite recent advances, evaluation for LALMs remains heavily underspecified relative to real-world requirements: most lack true linguistic and cultural authenticity, while others fail to capture acoustic realism. To bridge this gap, we propose GlobeAudio, a multilingual and multicultural benchmark designed to evaluate naturalistic audio understanding. GlobeAudio consists of 5,637 multiple-choice questions across six typologically diverse languages, expertly crafted by native speakers grounded on naturally occurring audio. In order to do well, models must possess higher-level auditory reasoning skills and culturally grounded interpretation. We systematically evaluate representative closed-source and open-source LALMs, as well as cascaded ASR-LLM pipelines. Our experiments reveal substantial performance gaps under natural acoustic conditions, particularly for open-source models and low-resource languages. These findings highlight critical limitations of current LALMs and underscore the importance of naturalistic audio evaluation for future audio-language systems. GlobeAudio can be found at https://huggingface.co/datasets/iNLP-Lab/GlobeAudio .
Large Audio-Language Models (LALMs) have been widely used as judge models for the automatic evaluation of generated speech. However, prior approaches predominantly focus on holistic naturalness, leaving fine-grained paralinguistic distinctions underexplored. We introduce ParaPairAudioBench, a pairwise benchmark of 5,175 audio pairs across five paralinguistic dimensions: Style, Rate, Emphasis, Age, and Gender. Our experiments show that current LALM judges still lag behind human judgments by 32%p on average and exhibit severe calibration failures, particularly in Tie cases where the correct decision is to abstain. To further analyze lexical versus acoustic reliance, the benchmark includes both same-transcript and cross-transcript conditions. ParaPairAudioBench enables multi-dimensional, calibration-aware assessment of the reliability of LALM-as-a-Judge for paralinguistic speech evaluation.
Jisu Jeon, Seungyeon Jwa, Joosung Lee +6
Hongik University, South Korea · Seoul National University, South Korea · NAVER Cloud, South Korea +1