A universal audio representation should capture fine-grained speech cues and high-level semantics for environmental sounds and music in a single encoder. Existing encoders often excel in one domain but degrade in others. We propose UniWhisper, an efficient continual multi-task training framework that casts heterogeneous audio tasks into a unified instruction and answer format. This enables standard next-token training without task-specific heads and losses. We train it on 38k hours of public audio and assess the encoder using shallow MLP probes and k-nearest neighbors (kNN) on 20 tasks spanning speech, environmental sound, and music. UniWhisper reaches normalized weighted averages of 0.81 with MLP probes and 0.61 with kNN, compared to 0.64 and 0.46 for Whisper, while retaining strong speech performance.
Figures & tables
Figure 1: Normalized per-task performance of UniWhisper on our 20-task extended HEAR [ 14 ] spanning speech, environmental sound, and music. Full results are reported in Table 3 .
Type
Name
Task
Duration
General
AudioCaps [ 16 ]
Captioning
142.5 h
AudioSet [ 17 ]
Tagging
5.8k h
LAION-Audio [ 18 ]
Captioning
4.3k h
WavCaps [ 19 ]
Captioning
7.6k h
Speech
AISHELL-1 [ 20 ]
ASR & SID
178 h
GigaSpeech [ 21 ]
ASR
10k h
Table 1: Public datasets used for continual multi-task training, grouped by domain. Duration denotes the total audio hours used after filtering and deduplication.
Type
Task
Metric
Weight
MLP
kNN
Speech
ASV2015
Acc
2000
2000
CREMA-D
Acc
1116
1116
Fluent Speech Commands
Acc
2000
2000
LibriCount
Acc
1144
1144
LibriSpeech-100h
iWER
10000
-
Table 2: Evaluation tasks, metrics, and weights used to compute normalized weighted averages. Weights are proportional to the number of test examples.
Domain
Task
wav2vec 2.0
HuBERT
WavLM
BEATs
CLAP
Whisper
CLAP-U
Whisper-U
Whisper-U-3
Ours
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
Speech
ASV2015
94.0
95.8
95.1
88.5
97.4
94.6
91.5
81.0
95.1
71.2
97.5
90.3
92.3
82.9
98.3
95.0
94.7
90.6
99.0
92.9
CREMA-D
55.5
21.8
64.5
34.0
71.3
28.0
66.5
39.7
36.9
25.4
57.5
38.2
54.3
49.0
64.3
17.8
66.3
30.0
84.4
67.9
FSC
47.2
1.6
96.5
3.6
97.5
4.3
47.3
5.5
3.6
1.6
97.8
23.4
17.7
5.4
97.2
1.0
98.0
53.1
82.7
14.6
LibriCount
58.0
20.8
58.2
18.2
65.1
38.8
68.2
37.8
38.8
28.1
59.4
45.2
60.9
54.0
62.8
9.1
53.7
41.4
64.4
47.2
LS-100h
16.0
–
83.4
–
64.1
–
15.8
–
0.2
–
62.5
–
1.5
–
79.0
–
89.9
–
91.6
–
Table 3: Per-task normalized results for MLP and kNN on 20 tasks. Scores are normalized to [0,1] using task-specific bounds so higher is better and reported as percentages (×100). CLAP-U replaces the Whisper encoder with CLAP-HASAT under the same training recipe. Whisper-U and Whisper-U-3 use the original Whisper Large-v3 decoder and train with one pass and three-pass replay over the same data, respectively. Highest score is in bold .
Task
Whisper
UniWhisper
CLAP-Uni
Whisper-Uni-1
Whisper-Uni-3
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
MLP
kNN
ASV2015
± 1.7
± 1.4
± 2.0
± 1.8
± 1.1
± 1.3
± 2.5
± 2.4
± 1.4
± 1.3
CREMA-D
± 0.9
± 0.8
± 1.4
± 1.2
± 0.9
± 1.1
± 1.7
± 0.3
± 1.3
± 0.4
FSC
± 1.8
± 0.4
± 1.6
± 0.2
± 0.4
± 0.1
± 1.8
± 0.3
± 2.5
± 0.7
LibriCount
± 0.9
± 0.8
± 1.0
± 0.9
± 1.5
± 0.6
± 1.6
± 0.2
± 1.2
± 0.8
LS-100h
± 1.0
–
± 1.8
–
± 0.3
–
± 1.9
–
± 1.8
–
Table 4: Reported deviations correspond to the maximum absolute difference between the mean and the endpoints of the 95% percentile CI, estimated using hierarchical bootstrap resampling over seeds and evaluation examples with replacement.
Audio-native large language models (audio-LLMs) commonly use Whisper as their audio encoder. However, Whisper was trained exclusively on speech data, producing weak representations for music and environmental sound. This forces downstream audio-LLMs to compensate through extensive training on large-scale non-speech data. We present Whisper-AuT, a domain-adapted audio encoder obtained by fine-tuning Whisper-large-v3 on a curated mixture of speech (80%), environmental sound (10%), and music (10%) totaling approximately 20M samples. The full encoder-decoder is trained end-to-end with a seq2seq captioning objective; the decoder is then discarded and only the encoder is retained. Linear probe evaluations show that Whisper-AuT achieves +23.0% on ESC-50 (environmental sound), +5.0% on GTZAN (music genre), and +0.7% on Speech Commands (keyword spotting) compared to the original Whisperlarge-v3 encoder. Whisper-AuT is designed as a drop-in replacement for Whisper in audio-LLM architectures, with the goal of reducing downstream training cost by providing stronger initial audio representations for non-speech domains.
Audio encoders are critical to modern audio applications as large language models (LLMs) increasingly rely on a single encoder for diverse inputs. While self-supervised learning (SSL) has yielded strong domain-specific encoders like speech or music experts, multi-domain approaches like USAD and SPEAR remain limited in coverage and evaluation. Recent studies also suggest supervised encoders align better with audio LLMs. We present USAD 2.0, a universal encoder integrating knowledge from both SSL and supervised foundation models. USAD 2.0 introduces domain-aware distillation to address teacher mismatch, extends coverage to the music domain, and adds second-stage supervised distillation for downstream use. We further scale the model to one billion parameters via depth scaling. Experiments show USAD 2.0 achieves strong or state-of-the-art performance across probing and LLM-based evaluations.
Heng-Jui Chang, Alexander H. Liu, Saurabhchand Bhati +4
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.