Organizations: Center of Auditory and Language Intelligence, Soochow University, Suzhou, China · School of Future Science and Engineering, Soochow University, Suzhou, China
Changes in listener acoustics require active noise control (ANC) filters to be redesigned for new acoustic paths. We introduce PRIME-ANC, a shared neural synthesizer that learns a bounded, path-dependent logmagnitude correction to a regularized path-ratio base. Minimum-phase reconstruction and truncation produce finite-impulse-response (FIR) filters; training optimizes their noise-control performance. Across ten random training/test splits per dataset, PRIME-ANC achieves average held-out one-third-octave reductions of 18.81 and 17.76 dB over 50 Hz-5 kHz on a ten-path dataset and a public earphone database, respectively. On original earphone measurements, it improves upon the path-ratio base by 7.89 dB. Ablation studies support the contributions of both the analytic base and the path-dependent correction. Given calibrated paths for a held-out listener condition, PRIME-ANC generates a path-specific FIR without iterative optimization. With three Gauss-Newton updates, it reaches 21.43 dB reduction, approaching direct weighted least-squares design while producing lower amplification and root-mean-square control output.
Figures & tables
Figure 1: PRIME-ANC: path-conditioned FIR synthesis from the magnitude base Abase=∣Wreg∣ and a shared neural generator.
Figure 2: Frequency-resolved NR (mean ±1 split SD). Gray: 40-s weighted FxLMS per path/noise (PANDAR: original measurements).
Figure 3: Common GN updates (a,b) and PANDAR core design time (c). Shading: ±1 split SD.
Method
Ten-path 3/7
PANDAR 7/16
PRIME-ANC, no interpolation
16.72±2.08
16.22±0.66
PRIME-ANC, paired interpolation
18.81±2.57
17.76±0.30
Ratio-base FIR (zero residual)
15.95±0.29
10.09±0.22
Projected complex-ratio FIR
5.95±0.63
7.09±0.28
FOMAML, pre-adaptation
9.43±0.65
9.35±0.22
Support-shared WMMSE
7.07±2.46
8.81±0.27
Table 1: No query updates: NR (dB), mean ± SD over ten splits.
Filter construction
NR
AMP
PRIME-ANC (standard)
17.755
0.134
Output construction controls
Ratio-base skip, MP
18.577
0.093
Ratio-base skip, MP + phase
18.616
0.093
No skip, MP
14.646
0.093
No skip, MP + phase
14.655
0.093
Table 2: PANDAR construction and conditioning: NR/AMP (dB).
Method
NR
AMP
RMS
Time (ms)
PRIME, direct
17.755
0.134
0.302
0.717
PRIME + GN(1)
20.869
0.136
0.303
1.317
PRIME + GN(2)
21.370
0.149
0.306
1.772
PRIME + GN(3)
21.430
0.156
0.307
2.169
Ratio-base + GN(4)
19.286
0.412
0.310
2.079
WMMSE, CPU direct
21.722
0.351
0.502
4.623
Table 3: PANDAR quality and core design time (NR/AMP in dB). WMMSE + GN starts from the CPU direct solution.
Traditional Active Noise Control (ANC) systems are mostly based on FxLMS algorithms, but such algorithms rely on linear assumptions and are often limited in handling broadband non-stationary noise or nonlinear acoustic paths. Not only that, the traditional method is used to eliminating all signals together, and noise reduction often accidentally damages the voice signal and affects normal communication. To tackle these issues, this study proposes a speech preserving deep learning ANC system, which aims to achieve stable noise reduction while effectively retaining speech in a complex acoustic environment. This study builds an end-to-end control architecture, the core of which adopts a Convolutional Recurrent Network (CRN). The structure uses the long short-term memory (LSTM) network to capture the time-related characteristics of acoustic signals. Combined with complex spectrum mapping (CSM) technology, the nonlinear distortion problem is effectively solved. In order to retain useful voice while removing noise, this study also designs a special voice retention loss function. This design guidance model selectively retains the target voice while suppressing environmental noise by identifying the characteristics of the spectrum structure. In addition, in order to verify whether the system is effective in real scenes, we use the Image Source Method (ISM) to build a high-fidelity acoustic simulation environment, which also simulates the real reverberation effect. Experimental results demonstrate that the proposed Deep ANC system achieves significantly better noise reduction than the traditional FxLMS algorithm, especially for non-stationary noises like crowd babble. Meanwhile, PESQ and STOI based evaluations confirm that the system preserves both the naturalness and intelligibility of the target speech.
Active noise cancellation (ANC) is widely deployed on consumer headphones and earbuds to suppress environmental noise. However, existing ANC systems require an error microphone at the user's ear canal to measure residual sound, preventing deployment on emerging open-ear wearable devices such as smart glasses and VR headsets, which leave the ear unoccluded. Here we present an ANC system for open-ear wearables that suppresses environmental noise using only microphones and miniaturized open-ear speakers embedded within the frame of the wearables, removing the need for an in-ear error microphone. Our low-latency computational pipeline uses a neural network to estimate the noise at the ear from an array of eight microphones distributed around the wearable's frame and generates an anti-noise signal in real-time. This mapping generalizes to unseen users and acoustic environments without prior acoustic measurement. We develop a custom glasses prototype and evaluate across eleven unseen users and eight unseen environments under mobility in the 100 to 1000 Hz frequency range, where environmental noise is concentrated. We achieve a mean noise reduction of 9.6 dB without any calibration, and 11.2 dB with a brief user-specific calibration. Further, we demonstrate that our approach extends to the broader class of open-ear wearables including VR headsets and headbands.
Kuang Yuan, Freddy Yifei Liu, Tong Xiao +5
Department of Electrical and Computer Engineering, Carnegie Mellon University, Pittsburgh, PA, USA · Department of Medical Physics and Acoustics, Carl von Ossietzky Universität Oldenburg, Germany · College of Computer Science and Technology, Zhejiang University, Hangzhou, China
A multi-task learning framework is proposed for optimizing a single deep neural network (DNN) for joint noise reduction (NR) and hearing loss compensation (HLC). A distinct training objective is defined for each task, and the DNN predicts two time-frequency masks. During inference, the amounts of NR and HLC can be adjusted independently by exponentiating each mask before combining them. In contrast to recent approaches that rely on training an auditory-model emulator to define a differentiable training objective, we propose an auditory model that is inherently differentiable, thus allowing end-to-end optimization. The audiogram is provided as an input to the DNN, thereby enabling listener-specific personalization without the need for retraining. Results show that the proposed approach not only allows adjusting the amounts of NR and HLC individually, but also improves objective metrics compared to optimizing a single training objective. It also outperforms a cascade of two DNNs that were separately trained for NR and HLC, and shows competitive HLC performance compared to a traditional hearing-aid prescription. To the best of our knowledge, this is the first study that uses an auditory model to train a single DNN for both NR and HLC across a wide range of listener profiles.
Philippe Gonzalez, Vera Margrethe Frederiksen, Torsten Dau +1
The authors are with the Department of Health Technology, Technical University of Denmark, 2800 Lyngby, Denmark.