Organizations: MiLM Plus, Xiaomi Inc., Beijing, China · DataoceanAI Inc., USA · Centre for Vision Speech and Signal Processing, University of Surrey, Guildford, UK · Shenzhen International Graduate School, Tsinghua University, Shenzhen, China
This paper presents the Interspeech 2026 Audio Encoder Capability Challenge, a benchmark specifically designed to evaluate and advance the performance of pre-trained audio encoders as front-end modules for Large Audio Language Models (LALMs). While LALMs have shown remarkable understanding of complex acoustic scenes, their performance depends on the semantic richness of the underlying audio encoder representations. This challenge addresses the integration gap by providing a unified generative evaluation framework, XARES-LLM, which assesses submitted encoders across a diverse suite of downstream classification and generation tasks. By decoupling encoder development from LLM fine-tuning, the challenge establishes a standardized protocol for general-purpose audio representations that can effectively be used for the next generation of multimodal language models.
Figures & tables
Figure 1: The XARES-LLM framework. Participants provide a frozen encoder, and the framework trains an LLM.
Track
Objective
LLM Decoder
A
Classification
Dedicated
A (Hidden)
B
Understanding
Dedicated
B (Hidden)
Shared with Track B
Table 1: Overview of Challenge Tasks and Model Training
ID
Dataset
Task Type
Metric
#
P1
ASV2015 [ 5 ]
Spoofing
Acc
2
P2
CREMA-D [ 6 ]
Emotion
Acc
5
P4
Fluent Spch Comm [ 7 ]
Intent
Acc
248
P5
LibriSpeech [ 8 ]
Gender
Acc
2
P9
LibriCount [ 9 ]
Speaker count
Acc
11
P11
Speech Commands [ 10 ]
Keyword spotting
Acc
30
Table 2: Track A (Classification) datasets in XARES-LLM, including mapping to Leaderboard IDs ( Pn,Hn ). Domains: blue (speech), yellow (sound), and red (music).
ID
Dataset
Task Type
Metric
U1
AISHELL-1-100h [ 23 ]
ASR
iCER
U2
Clotho [ 24 ]
Sound Caption
FENSE
U3
LibriSpeech-100h [ 8 ]
ASR
iWER
U4
MECAT [ 25 ]
General Caption
DATE
U5
The Song Describer Dataset [ 26 ]
Music Caption
FENSE
Hidden Evaluation Datasets
Table 3: Track B (Understanding) datasets in XARES-LLM, including hidden evaluation tasks. Public versions are hosted on Huggingface. Domains: blue (speech), yellow (sound), and red (music).
Team
Avg.
Public Test Results ( Pn )
Hidden Test Results ( Hn )
P1
P2
P3
P4
P5
P6
P7
P8
P9
P10
P11
P12
P13
P14
P15
H1
H2
H3
H4
H5
H6
THUVoice
91.2
99.0
87.9
91.3
99.4
95.9
32.1
88.4
90.9
47.9
78.8
93.5
88.9
93.9
97.2
93.9
87.5
99.9
98.5
98.1
100.0
99.0
THU-HSCI-2
90.8
98.6
86.0
92.3
99.4
89.1
33.9
89.1
93.9
50.6
78.7
82.6
89.6
94.5
97.7
87.3
87.5
99.7
99.5
98.8
100.0
83.0
trans-encoder
90.0
98.3
87.3
92.3
99.6
73.1
32.4
87.6
92.9
50.9
75.6
84.6
88.1
93.9
98.5
94.1
87.5
95.7
99.5
98.4
100.0
98.7
IASP Lab
89.4
99.0
85.2
91.5
99.2
71.6
30.9
85.9
89.9
49.3
75.3
81.0
84.8
94.0
98.3
97.5
86.4
99.4
99.5
97.4
100.0
96.7
MIT SLS
88.4
96.7
69.0
91.3
99.5
64.4
27.5
84.2
90.9
43.6
72.8
79.1
88.1
93.7
97.1
95.5
87.5
99.8
99.0
98.0
100.0
98.7
Table 4: Track A leader leaderboard results. Public ( P ) and Hidden ( H ) metrics are provided.
Team
Strategy
Pre-trained Encoders
Track A Rank
Track B Rank
THU-Voice
Mixture-of-Experts (MoE)
Audio-Flamingo 3, Qwen2-Audio
1
4
THU-HSCI-2
Dual-Model Fusion
Qwen2-Audio, Audio-Flamingo 3
2
3
trans-encoder
Hierarchical Extraction
StepAudio 2
3
1
IASP Lab
Dual-Branch Time Fusion
Qwen2-Audio, Whisper
4
10
EncodexPOL
Audio-text alignment
Audio-Flamingo 3, Whisper
6
5
NTU_BIO
Gated Feature Fusion
Qwen3-ASR
15
14
Table 6: Challenge submissions, their approach and (if applicable) the utilized pre-trained encoders. Methods utilizing audio encoders trained with a LALM are highlighted in grey .
Figure 2: Comparison of team rankings across Track A and Track B.
Audio-language model evaluation tests whether systems understand speech, music, and environmental sound reliably, fairly, and with evidence grounded in audio.
Large Audio Language Models (LALMs) rely on effective audio encoders for multi-task performance. We introduce UniAE-MoE, a unified audio encoder designed to model cross-domain audio representations and achieve outstanding downstream understanding performance via a Mixture-of-Experts (MoE) architecture. Specifically, we explore mainstream audio encoders and integrate those from Qwen2-Audio and Audio-Flamingo 3, which demonstrate superior downstream capabilities. To facilitate effective model fusion, we improve our encoder using SwiGLU with shared experts to decouple encoder networks, and we further introduce a two-stage instruction-tuning strategy to better adapt the model to diverse downstream tasks. Moreover, we propose the task-specific data scaling (TSDS) technique to enhance UniAE-MoE's understanding capabilities. On the XARES-LLM benchmark, UniAE-MoE attains a score of 0.802, achieving state-of-the-art performance. It also delivers top-tier performance in the official Interspeech 2026 Audio Encoder Capability Challenge, further demonstrating robust generalization across diverse audio tasks. Together, these results validate the effectiveness of UniAE-MoE for unified audio understanding across speech, music, and general audio domains.
Audio-language pretraining (ALP) holds promise for learning general-purpose audio representation, yet remains underexplored. Crucially, there is no consensus on whether audio-language models can build effective general-purpose audio encoders, nor a systematic understanding of how pretraining objectives behave across diverse tasks and scales. We identify three key barriers: limited scale of audio-text corpora, limited coverage of audio attributes in existing caption corpora, and lack of systematic exploration and evaluation. To fill this gap, we present the first principled empirical study of ALP. We first introduce CaptionStew, a 10.7M caption dataset aggregating open-source audio-text corpora across multiple domains and captioning focuses. We then conduct the first comprehensive evaluation comparing contrastive and captioning objectives for learning audio representation across speech, music, and environmental sound tasks. Our results not only demonstrate that ALP yields competitive, transferable representations, but reveal critical trade-offs: contrastive learning offers superior data efficiency, while captioning exhibits better scalability. Furthermore, we find that the benefits of supervised initialization often diminish at larger scales, challenging common practices. By grounding these claims in empirical evidence, we establish a viable pathway toward general-purpose audio representation learning, guiding future research.
Wei-Cheng Tseng, Xuanru Zhou, Mingyue Huo +3
University of Texas at Austin · Tencent AI Lab Seattle · Zhejiang University +1
Multi-encoder fusion extends Large Audio-Language Models (LALMs) beyond speech-centric recognition, but selecting encoders via intuition or exhaustive search often introduces redundant representations and inflates an already constrained compute budget. We propose CUES (Correlation-gUided Encoder Selection), a lightweight heuristic that estimates complementarity through task- and category-level Pearson correlations between encoders' performance profiles, scoring a candidate set from single-encoder evaluations alone--without fusion training during selection. Evaluated on the XARES-LLM benchmark with a frozen SmolLM2-135M backbone (LoRA-adapted) via five-fold cross-validation, CUES consistently identifies the same configuration per track from held-out development splits alone, without using test data for selection. For the broad TrackA suite, CUES selects a cross-family trio (Whisper-medium, mHuBERT-147, and Dasheng-base), achieving a 4.3% relative gain over Whisper-medium (0.771 vs. 0.739). For TrackB text generation, it re-anchors on a focused, speech-only pair (mHuBERT-147 and WavLM-base-plus) and actively abstains from adding a divergent encoder, outperforming mHuBERT-147 by 6.3% (0.589 vs. 0.554). Rather than a failure to scale, this divergence is consistent with a diversity--interference trade-off that CUES navigates per track from correlation signals alone: across the evaluated pool, added cross-family diversity tends toward an inverted-U on broad audio tasks but toward steady degradation on text generation, which favors a focused, speech-anchored set.
Pei-Jun Liao, Hung-Shin Lee, Wenze Ren +3
Inst. of Information Science, Academia Sinica, Taiwan · Grad. Inst. of Electrical Engineering, National Taiwan University, Taiwan · Grad. Inst. of AI Interdisciplinary Applied Technology, National Taiwan Normal University, Taiwan +3