Berlin Chen

Prof. Dr. Berlin Chen


Research Focus & Metrics

Academic Career Length

12Years (2015-Present)

(Calculated once Scilit publication data is synced: current year − year of first publication + 1)

Representative Publications

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13 October 2020 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
YWYu-Te Wu
Yu-Te Wu
BCBerlin Chen
Berlin Chen
LSLi Su
Li Su
Multi-instrument automatic music transcription (AMT) is a critical but less investigated problem in the field of music information retrieval (MIR). With all the difficulties faced by traditional AMT research, multi-instrument AMT needs further investigation on high-level music semantic modeling, efficient training methods for multiple attributes, and a clear problem scenario for systematic performance evaluation. In this paper, we propose a multi-instrument AMT method, with signal processing techniques specifying pitch saliency, novel deep learning techniques, and concepts partly inspired from multi-object recognition, instance segmentation, and image-to-image translation in computer vision. The proposed method is flexible for all the sub-tasks in multi-instrument AMT, including multi-instrument note tracking, a task that has rarely been investigated before. State-of-the-art performance is also reported in the sub-task of multi-pitch streaming.

26 August 2024 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
BYBi-Cheng Yan
Bi-Cheng Yan
BCBerlin Chen
Berlin Chen
Automatic pronunciation assessment (APA) manages to quantify second language (L2) learners' pronunciation proficiency in a target language by providing fine-grained feedback with multiple aspect scores (e.g., accuracy, fluency, and completeness) at various linguistic levels (i.e., phone, word, and utterance). Most of the existing efforts commonly follow a parallel modeling framework, which takes a sequence of phone-level pronunciation feature embeddings of a learner's utterance as input and then predicts multiple aspect scores across various linguistic levels. However, these approaches neither take the hierarchy of linguistic units into account nor consider the relatedness among the pronunciation aspects in an explicit manner. In light of this, we put forward an effective modeling approach for APA, termed HierGAT, which is grounded on a hierarchical graph attention network. Our approach facilitates hierarchical modeling of the input utterance as a heterogeneous graph that contains linguistic nodes at various levels of granularity. On top of the tactfully designed hierarchical graph message passing mechanism, intricate interdependencies within and across different linguistic levels are encapsulated and the language hierarchy of an utterance is factored in as well. Furthermore, we also design a novel aspect attention module to encode relatedness among aspects. To our knowledge, we are the first to introduce multiple types of linguistic nodes into graph-based neural networks for APA and perform a comprehensive qualitative analysis to investigate their merits. A series of experiments conducted on the speechocean762 benchmark dataset suggests the feasibility and effectiveness of our approach in relation to several competitive baselines.

1 January 2024
Verified
Conference Paper
Open access
PDF file
TLTien-Hong Lo
Tien-Hong Lo
FCFu-An Chao
Fu-An Chao
TWTzu-I Wu
Tzu-I Wu
YSYao-Ting Sung
Yao-Ting Sung
BCBerlin Chen
Berlin Chen
Tien-Hong Lo, Fu-An Chao, Tzu-i Wu, Yao-Ting Sung, Berlin Chen. Findings of the Association for Computational Linguistics: NAACL 2024. 2024.

31 December 2021 in IEEE MultiMedia
Verified
Journal Article
Closed access
FCFu-An Chao
Fu-An Chao
JHJeih-weih Hung
Jeih-weih Hung
TSTommy Sheu
Tommy Sheu
BCBerlin Chen
Berlin Chen
Due to the enormous progress in deep learning, speech enhancement (SE) techniques have shown promising efficacy and play a pivotal role prior to an automatic speech recognition (ASR) system to mitigate the noise effects. By virtue of a wide variety of deep neural network (DNN) approaches, some researches have suggested that the phase information of speech signals is the key to success on SE. In this study, we present a continuation of our previous work and put forward a novel Cross-domain Time-reversal Enhancement NETwork (CD-TENET). CD-TENET leverages the time-reversed version of a speech signal and two features that consider the phase information of a speech signal in the time domain and the frequency domain, respectively, to promote SE performance for noise-robust ASR. Extensive experiments conducted on the Voicebank-DEMAND benchmark dataset show that CD-TENET can not only recover the original speech effectively but also improve both SE and ASR performance simultaneously. More surprisingly, the proposed CD-TENET method can offer a marked relative word error rate (WER) reduction of 43 % on the test set of scenarios contaminated with unseen noises when compared to a strong baseline with the multi-condition training setting.


9 May 2026 in Applied Sciences
Verified
Journal Article
Open access
PDF file
CCCHIEN CHIA CHIU
CHIEN CHIA CHIU
TCTai-You Chen
Tai-You Chen
TWTzu-Wei Wang
Tzu-Wei Wang
BCBerlin Chen
Berlin Chen
JHJeih-weih Hung
Jeih-weih Hung
Voice Activity Detection (VAD) and Personal Voice Activity Detection (PVAD) are fundamental components in modern voice-based human–machine interaction systems. While VAD distinguishes speech from non-speech segments, PVAD further identifies whether the detected speech belongs to a specific target speaker, enabling more robust performance in multi-speaker environments. Recently, the Flexible Dynamic Encoder RNN (FDE-RNN) has demonstrated state-of-the-art performance on PVAD tasks by leveraging a detachable Personalization module (P-module) built upon a Dynamic Encoder RNN backbone. However, the Long Short-Term Memory (LSTM) networks employed throughout FDE-RNN inherently suffer from sequential processing constraints that prevent parallelization across time steps, and their fixed-size hidden state may restrict representational capacity for fine-grained speaker discrimination. In this paper, we propose FDE-Mamba, which replaces all three LSTM components in FDE-RNN—the Prediction RNN, the Encoder RNN, and the P-module temporal model—with independent Mamba blocks, each equipped with a selective state space mechanism and an expansion layer for enriched feature representation. The proposed architecture retains the weighted residual connection, FiLM-based speaker embedding fusion, and parallel training strategy of the original FDE-RNN without modification. Experimental results on the LibriSpeech corpus demonstrate that FDE-Mamba achieves a PVAD mAP of 0.9605, representing a 1.97% improvement over the reproduced FDE-RNN baseline (0.9419), along with an accuracy improvement from 86.85% to 89.87% and a 3.16× reduction in real-time factor owing to the memory-efficient linear recurrences of the Mamba selective scan during inference, alongside its inherent parallelizability during training. Ablation studies further confirm that both the D skip connection and the expansion layer within each Mamba block contribute meaningfully to the observed performance gains, validating the effectiveness of each architectural design choice. These results suggest that Mamba is a compelling alternative to LSTM for temporal modeling in PVAD systems, and that the proposed integration provides a design blueprint for future selective SSM applications in gated PVAD architectures.