Target speech extraction (TSE) typically relies on pre-recorded high-quality enrollment speech, which disrupts user experience and limits feasibility in spontaneous interaction. In this paper, we propose Enroll-on-Wakeup (EoW), a novel framework where the wake-word segment, captured naturally during human-machine interaction, is automatically utilized as the enrollment reference. This eliminates the need for pre-collected speech to enable a seamless experience. We perform the first systematic study of EoW-TSE, evaluating advanced discriminative and generative models under real diverse acoustic conditions. Given the short and noisy nature of wake-word segments, we investigate enrollment augmentation using LLM-based TTS. Results show that while current TSE models face performance degradation in EoW-TSE, TTS-based assistance significantly enhances the listening experience, though gaps remain in speech recognition accuracy.
Figures & tables
Figure 1: Illustration of EoW-TSE system.
Scenario
Condition
#spk/#utt
Average duration (s)
enrollment
mixture
CloseNoise-10
d=1, RT=0.4
45/503
1.08
1.78
FarNoise-10
d=3, RT=0.4
45/502
1.08
1.78
FarNoiseReverb-10
d=3, RT=0.6
30/278
1.17
1.79
FarNoise-5
d=3, RT=0.4
45/503
1.08
1.78
FarNoiseReverb-5
d=3, RT=0.6
15/226
0.97
1.76
Table 1: Acoustic conditions of five EoW-TSE test scenarios. ‘ d ’ denotes distance to the microphone (m), ‘ RT ’ is the reverberation time ( RT60 in seconds), and ‘*-10/-5’ indicates SNR levels of 10dB and 5dB respectively.
Models
SI-SDR
PESQ
STOI
Params
MACs
Mixture
-1.96
1.08
64.73
-
-
SEF-PNet [ 9 ]
8.18
1.55
82.67
6.08M
15.87G
CIE-mDPTNet [ 10 ]
9.47
1.78
85.35
2.9M
48.3G
⋆ LExt [ 11 ]
10.47
1.88
87.26
3.9M
61.0G
SoloSpeech [ 18 ]
11.12
1.89
-
590.9M
-
Table 2: Results on Libri2Mix 2-Speaker+Noise. ⋆ are the results reproduced by us.
Conditions
DNSMOS ↑
WER ↓
Noisy
xTTS
IndexTTS2
CosyVoice3
Noisy
xTTS
IndexTTS2
CosyVoice3
CloseNoise-10
2.029
2.472
2.564
2.609
14.61
39.76
11.23
40.36
FarNoise-10
1.590
2.464
2.237
2.428
27.74
40.04
19.27
54.88
FarNoiseReverb-10
1.262
2.549
1.917
2.114
19.69
33.45
16.10
59.71
FarNoise-5
1.440
2.274
2.063
2.400
45.13
46.57
36.58
80.62
FarNoiseReverb-5
1.251
2.473
1.743
2.107
70.58
55.86
44.47
102.54
Table 3: Results on enrollments synthesized using different TTS models.
Conditions
DNSMOS ↑
WER ↓
Noisy
xTTS (CR)
IndexTTS2 (CR)
xTTS(EC)
IndexTTS2(EC)
Noisy
xTTS(CR)
IndexTTS2(CR)
xTTS(EC)
IndexTTS2(EC)
CloseNoise-10
1.904
2.163
2.119
2.197
2.170
3.10
4.37
3.24
3.91
3.48
FarNoise-10
1.371
1.511
1.481
1.527
1.500
4.54
6.41
5.38
6.81
5.14
FarNoiseReverb-10
1.165
1.202
1.200
1.214
1.195
3.52
7.15
5.06
7.00
6.08
FarNoise-5
1.286
1.379
1.366
1.396
1.371
13.31
16.01
29.55
15.92
25.61
FarNoiseReverb-5
1.173
1.216
1.223
1.217
1.212
15.95
22.89
31.32
23.12
28.36
Table 4: EoW-TSE results with different enrollment augmentation methods on CIE-mDPTNet.
Figure 2: OVRL scores on five scenarios: Original EoW-TSE vs. TTS-augmented (CR).
Figure 3: WERs on five scenarios: Original EoW-TSE vs. TTS-augmented (CR).
Target speech extraction (TSE) typically trains a separate extractor per cue, and visual-cue systems often need corruption-matched training to remain robust under visual frame corruption. We show that one autoregressive LLM backbone, TSE-Omni, can serve both temporally synchronous cues (lip movements, co-speech gestures) and asynchronous cues (enrollment audio, text). TSE-Omni is driven by next-token prediction: each step predicts target speech semantic tokens from its own past outputs, which we term self-enrollment, forming a continuous target-speech context initialized by the enrollment cue (asynchronous audio or text, or a short visual prefix). This enables audio-visual compensation: the model uses synchronized visuals when intact and its token history when visual frames are missing. Under clean visuals, TSE-Omni matches strong discriminative and generative baselines (SpeechBERTScore 0.81 on VoxCeleb2 and 0.89 on LRS3 zero-shot) with higher DNSMOS. On the same VoxCeleb2 test set, after a 2 s clean visual start, removing the remaining visual frames leaves SpeechBERTScore at 0.81. It remains usable under sparse overlap and multi-speaker interference, and supports streaming inference. Project page: https://alexwxwu.github.io/tseomni-main/.
Target speaker extraction (TSE) aims to recover a target speaker's speech from a mixture using a reference utterance as a cue. Most TSE systems adopt conditional auto-encoder architectures with one-step inference. Inspired by test-time scaling, we propose a training-free multi-step inference method that enables iterative refinement with a frozen pretrained model. At each step, new candidates are generated by interpolating the original mixture and the previous estimate, and the best candidate is selected for further refinement until convergence. Experiments show that, when ground-truth target speech is available, optimizing an intrusive metric (SI-SDRi) yields consistent gains across multiple evaluation metrics. Without ground truth, optimizing non-intrusive metrics (UTMOS or SpkSim) improves the corresponding metric but may hurt others. We therefore introduce joint metric optimization to balance these objectives, enabling controllable extraction preferences for practical deployment.
Zhenghai You, Ying Shi, Lantian Li +1
Beijing University of Posts and Telecommunications, China · Center for Speech and Language Technologies, BNRist, Tsinghua University, China
Recent ASR development has placed growing emphasis on generalization across diverse domains and acoustic conditions. Existing approaches typically adapt pretrained ASR models to front-end functions such as wake-up word (WuW) detection through additional training or task-specific modules. In this work, we explore the use of a shared pretrained ASR backbone for WuW detection without gradient-based fine-tuning and examine whether a compact encoder can be extracted using the PCA-based structured pruning approach of SliceGPT. Experiments with Parakeet-TDT-0.6B-v3 and Moonshine-base show that WuW detection performance remains relatively stable when the encoder channel dimension is reduced by 50%. These results suggest that task-relevant compact encoders can be derived from pretrained ASR models without fine-tuning.