Audio language models process input audio into rich frame-level representations, but the standard approach to temporal localization generates timestamps as sequences of text tokens, which discards the frame-level representations in favor of autoregressive decoding. However, generating timestamps as tokens is slow and not parallelizable, and tends to hallucinate when producing timestamps outside the training distribution. We propose internal frame-level reuse, a method that trains audio LMs to reuse their own internal audio representations for temporal localization directly, bypassing token generation altogether. We introduce a lightweight prediction head trained via different frame-level objectives: a binary frame classifier and a novel inhomogeneous Poisson process (IHP) loss that models temporal event intensity. Across word localization, speaker diarization, and event localization tasks, our approach can achieve a >50x inference speedup over token-based generation and demonstrates robust length generalization, maintaining high accuracy on out-of-distribution audio durations where token-based models collapse completely. We find that reusing audio frame-level representations yields localization accuracies comparable to (and often better than) finetuned token-based baselines.
Figures & tables
Figure 1: Frame-level prediction versus standard autoregressive generation for predicting timestamps in audio. At inference time, we extract the predicted timestamp from the binary frame-level probabilities p1,..,pT via argmax.
Figure 2: Inhomogeneous Poisson process frame-level training. In the simple case where only 1 timestamp is needed, the logλ1,...,logλT are used to construct a non-parametric probability density estimator over the audio frames, normalized by the total intensity Λ(T)=∑k=1Tλk .
Figure 3: Speed-up factor between token-based and Poisson-based timestamp generation across batch sizes. Efficiency gains are larger when the timestamp sequences are long. Poisson frame-level timestamps are as much 60× faster than token-based timestamps.
Train
Test
Tokens-only
Poisson
Binary
20ms ↑
40ms ↑
MAD ↓
20ms ↑
40ms ↑
MAD ↓
20ms ↑
40ms ↑
MAD ↓
Matched Distributions
0-4s
0-4s
93.5%
97.5%
0.01
94.1%
98.0%
0.01
85.4%
96.3%
0.03
0-8s
0-8s
93.7%
97.6%
0.01
93.9%
98.1%
0.01
88.2%
96.9%
0.02
0-12s
0-12s
93.5%
97.8%
0.01
94.1%
98.3%
0.01
86.6%
95.9%
0.03
0-16s
0-16s
93.7%
97.9%
0.01
93.9%
98.4%
0.01
85.3%
96.2%
0.04
Table 1: Librispeech word alignment accuracies and mean absolute deviation (MAD, in seconds) under matched (e.g., train on timestamps between 0–8 seconds, test on timestamps that are between 0–8 seconds) and mismatched (e.g., train on 0-8 seconds but test on 8–12 seconds) settings. The Poisson loss generalizes to out-of-distribution timestamps, unlike the token loss.
Librispeech
Libricount
AudioSet
Model
Method
20ms ↑
40ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
Qwen2.5 3B
Tokens
91.4%
96.0%
0.08
60.8%
73.2%
0.13
40.9%
55.1%
1.15
Poisson
93.8%
97.9%
0.01
64.8%
75.2%
0.13
41.0%
59.3%
1.02
Binary
85.8%
96.1%
0.02
58.4%
72.0%
0.17
41.0%
57.7%
1.08
Qwen2.5 7B
Tokens
93.2%
97.1%
0.03
62.1%
75.3%
0.13
40.1%
56.9%
1.03
Poisson
93.3%
98.1%
0.01
65.9%
76.5%
0.12
42.3%
58.5%
0.99
Table 2: Finetuned accuracies (Acc, %) and MAD (seconds) for Qwen audio LLMs with and without frame-level reuse. Highest accuracies and lowest MADs are bolded. Models tuned with Poisson loss outperform models tuned with either token or binary loss.
# of Words
Tokens-only
Poisson
20ms ↑
40ms ↑
MAD ↓
20ms ↑
40ms ↑
MAD ↓
1-5
94.6%
98.1%
0.01
94.0%
97.4%
0.01
6-10
94.2%
97.3%
0.01
94.9%
97.8%
0.01
11-15
94.7%
97.8%
0.01
95.3%
98.1%
0.01
16-20
94.3%
97.3%
0.01
94.5%
97.1%
0.01
21-25
94.5%
97.6%
0.01
94.4%
97.3%
0.01
Table 3: Qwen2.5-Omni 7B performance on predicting start-of-word timestamps in Librispeech, stratified by number of words in the transcript. Training with either the token or Poisson loss result in very similar performance.
Librispeech (Word Align.)
Libricount (Speaker Diar.)
Audioset (Human Vocal.)
Model
20ms ↑
40ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
Audio Flamingo 3
0.6%
1.4%
1.82
2.3%
6.3%
1.53
5.8%
10.2%
1.69
Voxtral 3B
0.8%
1.1%
2.52
0.3%
1.2%
6.02
1.2%
2.2%
3.74
Voxtral 24B
0.2%
0.5%
2.12
0.4%
1.4%
4.91
2.3%
4.2%
4.03
Qwen2.5 3B
0.1%
0.2%
2.59
0.6%
1.9%
3.71
3.6%
4.5%
4.46
Qwen2.5 7B
0.7%
1.1%
2.17
0.4%
1.5%
5.29
3.9%
5.7%
3.52
Table 4: Zero-shot performance of audio LMs on temporal tasks. We report accuracies (%) at different tolerances (20ms, 40ms, 100ms) and mean absolute deviation (MAD, seconds). Models are prompted to generate timestamps directly from audio without task-specific fine-tuning. Results show that all models struggle in this zero-shot setting, with low accuracy and high MAD across tasks.
Librispeech
Libricount
AudioSet
20ms ↑
40ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
Tokens-only
91.4%
96.0%
0.08
60.8%
73.2%
0.13
40.9%
55.1%
1.15
Poisson
93.8%
97.9%
0.01
64.8%
75.2%
0.13
41.0%
59.3%
1.02
Interpolated loss with Poisson frame-level loss and token loss
Token Output
92.2%
96.4%
0.06
62.8%
72.7%
0.15
41.2%
57.5%
1.01
Poisson Output
94.2%
98.1%
0.01
63.7%
74.0%
0.15
41.3%
58.5%
1.04
Table 5: Comparison of Qwen2.5-Omni 3B model performance between using token loss alone, Poisson loss alone, and an interpolation of the two losses. We ran all of our experiments with Linterpolated=Ltokens+0.05LPoisson . Interpolating the two losses shows no consistent improvement on token outputs.
Appendix figures & tables4 assets
Supplementary material from the paper’s appendix.
Appendix
Hyperparameter
Value
Learning Rate
1×10−6
Effective Batch Size
8
Optimizer
AdamW
Adam β1,β2
0.9, 0.999 (default)
Weight Decay
0.01 (default)
Training Epochs
Appendix
Table 6: Hyperparameter settings and training configurations.
Models
Sound
Music
Speech
Average
Qwen2-Audio-Instruct 8B
45.9
53.3
45.9
52.5
Qwen2.5-Omni 7B (Base Model)
64.7
61.2
64.1
63.3
Qwen2.5-Omni 7B (Poisson-Finetuned)
63.9
56.9
61.4
60.7
Appendix
Table 7: MMAU results on Sound, Music, Speech, and their average on the full test set.
Librispeech
Libricount
AudioSet
Model
20ms ↑
40ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
40ms ↑
100ms ↑
MAD ↓
3B
0.08
0.11
2.33
0.18
0.23
1.55
0.05
0.11
2.97
7B
0.04
0.06
2.21
0.14
0.19
1.79
0.10
0.19
2.74
Appendix
Table 8: Performance of the 3B and 7B models on Librispeech, Libricount, and AudioSet at different temporal resolutions when the audio LM is frozen and only the linear prediction head is trained.
Figure 4: Multi-timestamp decoding with our iterative mode finding and smoothing heuristic.
National Engineering Research Center of Speech and Language Information Processing, University of Science and Technology of China, Hefei, China · ICT Cluster, Singapore Institute of Technology, Singapore