We propose a modular speaker diarization framework that collaborates with front-end techniques in a three-stage process, designed for the challenging CHiME-8 NOTSOFAR-1 acoustic environment. The framework leverages the strengths of deep learning based speech separation systems and traditional speech signal processing techniques to provide more accurate initializations for the Neural Speaker Diarization (NSD) system at each stage, thereby enhancing the performance of a single-channel NSD system. Firstly, speaker overlap detection and Continuous Speech Separation (CSS) are applied to the multichannel speech to obtain clearer single-speaker speech segments for the Clustering-based Speaker Diarization (CSD), followed by the first NSD decoding. Next, the binary speaker masks from the first decoding are used to initialize a complex Angular Center Gaussian Mixture Model (cACGMM) to estimate speaker masks on the multi-channel speech. Using Mask-to-VAD post-processing techniques, we achieve per-speaker speech activity with reduced speaker error (SpkErr), followed by a second NSD decoding. Finally, the second decoding results are used to Guide Source Separation (GSS) to produce per-speaker speech segments. Short utterances containing one word or fewer are filtered, and the remaining speech segments are re-clustered for the final NSD decoding. We present evaluation results progressively explored from the CHiME-8 NOTSOFAR-1 challenge, demonstrating the effectiveness of our modular diarization system and its contribution to improving speech recognition performance. The code will be open-sourced at https://github.com/rywang99/USTC-NERCSLIP_CHiME-8 . • We propose a novel three-stage modular speaker diarization framework integrating front-end cues. • CSS streams extend single-speaker segments for CSD clustering and NSD decoding initialization. • Spatial information is utilized to progressively reduce SpkErr and improve ASR performance.

Although fully end-to-end speaker diarization systems have made significant progress in recent years, modular systems often achieve superior results in real-world scenarios due to their greater adaptability and robustness. Historically, modular speaker diarization methods have seldom discussed how to leverage spatial cues from multi-channel speech. This paper proposes a three-stage modular system to enhance single-channel neural speaker diarization systems and recognition performance by utilizing spatial cues from multi-channel speech to provide more accurate initialization for each stage of neural speaker diarization (NSD) decoding: (1) Overlap detection and continuous speech separation (CSS) on multi-channel speech are used to obtain cleaner single speaker speech segments for clustering, followed by the first NSD decoding pass. (2) The results from the first pass initialize a complex Angular Central Gaussian Mixture Model (cACGMM) to estimate speaker-wise masks on multi-channel speech, and through Overlap-add and Mask-to-VAD, achieve initialization with lower speaker error (SpkErr), followed by the second NSD decoding pass. (3) The second decoding results are used for guided source separation (GSS), recognizing and filtering short segments containing less one word to obtain cleaner speech segments, followed by re-clustering and the final NSD decoding pass. We presented the progressively explored evaluation results from the CHiME-8 NOTSOFAR-1 (Natural Office Talkers in Settings Of Far-field Audio Recordings) challenge, demonstrating the effectiveness of our system and its contribution to improving recognition performance. Our final system achieved the first place in the challenge.

Text-to-music (TTM) generation, which converts textual descriptions into audio, opens up innovative avenues for multimedia creation. Achieving high quality and diversity in this process demands extensive, high-quality data, which are often scarce in available datasets. Most open-source datasets frequently suffer from issues like low-quality waveforms and low text-audio consistency, hindering the advancement of music generation models. To address these challenges, we propose a novel quality-aware training paradigm for generating high-quality, high-musicality music from large-scale, quality-imbalanced datasets. Additionally, by leveraging unique properties in the latent space of musical signals, we adapt and implement a masked diffusion transformer (MDT) model for the TTM task, showcasing its capacity for quality control and enhanced musicality. Furthermore, we introduce a three-stage caption refinement approach to address low-quality captions' issue. Experiments show state-of-the-art (SOTA) performance on benchmark datasets including MusicCaps and the Song-Describer Dataset with both objective and subjective metrics. Demo audio samples are available at https://qa-mdt.github.io/, code and pretrained checkpoints are open-sourced at https://github.com/ivcylc/OpenMusic.

Advanced Audio- Visual Speech Recognition (AVSR) sys-tems have been observed to be sensitive to missing video frames, performing even worse than single-modality mod-els. While applying the common dropout techniques to the video modality enhances robustness to missing frames, it simultaneously results in a performance loss when dealing with complete data input. In this study, we delve into this contrasting phenomenon through the lens of modality bias and uncover that an excessive modality bias towards the audio modality induced by dropout constitutes the fun-damental cause. Next, we present the Modality Bias Hy-pothesis (MBH) to systematically describe the relationship between the modality bias and the robustness against missing modality in multimodal systems. Building on these findings, we propose a novel Multimodal Distribution Approxi-mation with Knowledge Distillation (MDA-KD)framework to reduce over-reliance on the audio modality, maintaining performance and robustness simultaneously. Finally, to address an entirely missing modality, we adopt adapters to dynamically switch decision strategies. The effective-ness of our proposed approach is evaluated through comprehensive experiments on the MISP2021 and MISP2022 datasets. Our code is available at https://github.com/dalision/ModalBiasAV5R.

Deep learning (DL)-based speaker diarization methods have proven powerful performance comparing to traditional clustering-based methods for multi-talker speech diarization and recognition in farfield scenes. However, most DL-based approaches cannot utilize the spatial information well due to the poor robustness to unknown array topology and acoustic scenario. In this paper, a spatial long-term iterative mask estimation (SLT-IME) method is proposed to improve the performance of speaker diarization in various real-world acoustic scenarios. First, the complex angular central gaussian mixture model (cACGMM) with diarization results as initial values is used to estimate the presence probability of each speaker at each time-frequency bin, namely speaker masks, in a long-term chunk. Then, the speaker masks are converted to speaker activities according to the threshold, which deliver the diarization information of which speaker is active and when. Finally, the estimated speaker activity can also serve as the initial input for the diarization system, resulting in improved ASR performance. Experimental results on the CHiME-7 three datasets (CHiME-6, DiPCo, Mixer 6) show proposed method can improve diarization and recognition systems performance simultaneously. It also plays a key role in the ensemble system that achieves the best performance in the main track of CHiME-7 DASR Challenge.

In multi-speaker scenarios, automatic speech recognition (ASR) models rely on pre-processed audio after speaker separation. However, when the target speaker is not accurately separated, ASR models face limitations in reaching their peak performance. To address this issue, we propose a speaker-adaptive ASR framework that possesses more implicit target speaker enhancement capability by efficiently joint-optimized speaker recognition (SR) and ASR models. Our framework introduces sharing self-supervised learning representation, optimization transfer and hierarchy speaker-gated attention. In this manner, it can maximize effectiveness of embedding bias and emphasize target speaker corresponding to semantic units. In the CHiME-7 DASR sub-track, the proposed method achieves a 28.19% relative reduction in word error rate (WER) on the development sets when compared to the official baseline. Notably, this framework has also been employed in the champion system for the CHiME-7 DASR.

Previous Multimodal Information based Speech Processing (MISP) challenges mainly focused on audio-visual speech recognition (AVSR) with commendable success. However, the most advanced back-end recognition systems often hit performance limits due to the complex acoustic environments. This has prompted a shift in focus towards the Audio-Visual Target Speaker Extraction (AVTSE) task for the MISP 2023 challenge in ICASSP 2024 Signal Processing Grand Challenges. Unlike existing audio-visual speech enhancement challenges primarily focused on simulation data, the MISP 2023 challenge uniquely explores how front-end speech processing, combined with visual clues, impacts back-end tasks in real-world scenarios. This pioneering effort aims to set the first benchmark for the AVTSE task, offering fresh insights into enhancing the accuracy of back-end speech recognition systems through AVTSE in challenging and real acoustic environments. This paper delivers a thorough overview of the task setting, dataset, and baseline system of the MISP 2023 challenge. It also includes an in-depth analysis of the challenges participants may encounter. The experimental results highlight the demanding nature of this task, and we look forward to the innovative solutions participants will bring forward.

We propose a novel neural speaker diarization system using memory-aware multi-speaker embedding with sequence-to-sequence architecture (NSD-MS2S), which integrates the strengths of memory-aware multi-speaker embedding (MA-MSE) and sequence-to-sequence (Seq2Seq) architecture, leading to improvement in both efficiency and performance. Next, we further decrease the memory occupation of decoding by incorporating input features fusion and then employ a multi-head attention mechanism to capture features at different levels. NSD-MS2S achieved a macro diarization error rate (DER) of 15.9% on the CHiME-7 EVAL set, which signifies a relative improvement of 49% over the official baseline system, and is the key technique for us to achieve the best performance for the main track of CHiME-7 DASR Challenge. Additionally, we introduce a deep interactive module (DIM) in MA-MSE module to better retrieve a cleaner and more discriminative multi-speaker embedding, enabling the current model to outperform the system we used in the CHiME-7 DASR Challenge. Our code is available at https://github.com/liyunlongaaa/NSD-MS2S.

The development of quantum machine learning demonstrates its quantum advantages over traditional deep learning, which promises to discover new patterns on supervised classification datasets. This work proposes a classical-to-quantum transfer learning system based on the large-scale unsupervised pre-trained model to demonstrate the competitive performance of quantum transfer learning for synthetic speech detection. We use the pre-trained model WavLM-Large to extract feature maps from speech signals, obtain low-dimensional embedding vectors through classical network components, and then jointly fine-tune the pre-trained model and classical network components with a variational quantum circuit (VQC). We evaluate our system on the ASVspoof 2021 DF task, and the experiments using quantum circuit simulations show that quantum transfer learning can improve the performance of the classical transfer learning baseline on the task.