Ziqiang Zhang

Ziqiang Zhang

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Ziqiang Zhang

National Engineering Research Center of Speech and Language Information Processing, University of Science and Technology of China, Hefei, China

Research Focus & Metrics

Academic Career Length

5Years (2022-Present)

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

Representative Publications

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15 January 2025 in IEEE Transactions on Audio, Speech and Language Processing
Verified
Journal Article
Closed access
SCSanyuan Chen
Sanyuan Chen
CWChengyi Wang
Chengyi Wang
YWYu Wu
Yu Wu
ZZZiqiang Zhang
Ziqiang Zhang
LZLong Zhou
Long Zhou
FWFuru Wei
Furu Wei
We introduce a language modeling approach for text to speech synthesis (TTS). Specifically, we train a neural codec language model (called VALL-E) using discrete codes derived from an off-the-shelf neural audio codec model, and regard TTS as a conditional language modeling task rather than continuous signal regression as in previous work. During the pre-training stage, we scale up the TTS training data to 50k hours of English speech which is hundreds of times larger than existing systems. VALL-E emerges in-context learning capability and can be used to synthesize high-quality personalized speech with only a 3-second enrolled recording of an unseen speaker as a prompt. Experiment results show that VALL-E significantly outperforms the state-of-the-art zero-shot TTS system in terms of speech naturalness and speaker similarity. In addition, we find VALL-E could preserve the speaker's emotion and acoustic environment from the prompt in synthesis.

20 March 2024 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
ZZZiqiang Zhang
Ziqiang Zhang
SCSanyuan Chen
Sanyuan Chen
LZLong Zhou
Long Zhou
YWYu Wu
Yu Wu
SRShuo Ren
Shuo Ren
FWFuru Wei
Furu Wei
How to boost speech pre-training with textual data is an unsolved problem due to the fact that speech and text are very different modalities with distinct characteristics. In this paper, we propose a cross-modal Speech and L anguage M odel ( SpeechLM ) to explicitly align speech and text pre-training with a pre-defined unified discrete representation. Specifically, we introduce two alternative discrete tokenizers to bridge the speech and text modalities, including phoneme-unit and hidden-unit tokenizers, which can be trained using unpaired speech or a small amount of paired speech-text data. Based on the trained tokenizers, we convert the unlabeled speech and text data into tokens of phoneme units or hidden units. The pre-training objective is designed to unify the speech and the text into the same discrete semantic space with a unified Transformer network. We evaluate SpeechLM on various spoken language processing tasks including speech recognition, speech translation, and universal representation evaluation framework SUPERB, demonstrating significant improvements on content-related tasks. Code and models are available at https://aka.ms/SpeechLM.

12 May 2023 in IEEE Transactions on Multimedia
Verified
Journal Article
Closed access
QZQiushi Zhu
Qiushi Zhu
LZLong Zhou
Long Zhou
ZZZiqiang Zhang
Ziqiang Zhang
SLShujie Liu
Shujie Liu
BJBinxing Jiao
Binxing Jiao
FWFuru Wei
Furu Wei
Although speech is a simple and effective way for humans to communicate with the outside world, a more realistic speech interaction contains multimodal information, e.g., vision, text. How to design a unified framework to integrate different modal information and leverage different resources (e.g., visual-audio pairs, audio-text pairs, unlabeled speech, and unlabeled text) to facilitate speech representation learning was not well explored. In this paper, we propose a unified cross-modal representation learning framework VatLM (Visual-Audio-Text Language Model). The proposed VatLM employs a unified backbone network to model the modality-independent information and utilizes three simple modality-dependent modules to preprocess visual, speech, and text inputs. In order to integrate these three modalities into one shared semantic space, VatLM is optimized with a masked prediction task of unified tokens, given by our proposed unified tokenizer. We evaluate the pre-trained VatLM on audio-visual related downstream tasks, including audio-visual speech recognition (AVSR), and visual speech recognition (VSR) tasks. Results show that the proposed VatLM outperforms previous state-of-the-art models, such as the audio-visual pre-trained AV-HuBERT model, and analysis also demonstrates that VatLM is capable of aligning different modalities into the same space. To facilitate future research, we release the code and pre-trained models at https://aka.ms/vatlm .

10 May 2023 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
QZQiu-Shi Zhu
Qiu-Shi Zhu
Jie ZhangJie Zhang
Jie Zhang
ZZZi-Qiang Zhang
Zi-Qiang Zhang
LDLi-Rong Dai
Li-Rong Dai
Though speech enhancement (SE) can be used to improve speech quality in noisy environments, it may also cause distortions that degrade the performance of automatic speech recognition (ASR) models. Self-supervised pre-training, on the other hand, has been shown to improve the noise robustness of ASR models. However, the potential of the (optimal) integration of SE and self-supervised pre-training still remains unclear. In this paper, we propose a novel self-supervised pre-training framework that incorporates SE to improve ASR performance in noisy environments. First, in the pre-training phase the original noisy waveform or the waveform obtained by SE is fed into the self-supervised model to learn the contextual representation, where the quantized clean speech acts as the target. Second, we propose a dual-attention fusion method to fuse the features of noisy and enhanced speech, which can compensate for the information loss caused by separately using individual modules. Due to the flexible exploitation of clean/noisy/enhanced branches, the proposed method turns out to be a generalization of some existing noise-robust ASR models, e.g., enhanced wav2vec2.0. Finally, experimental results on both synthetic and real noisy datasets show that the proposed joint training approach can improve the ASR performance under various noisy settings, leading to a stronger noise robustness.

29 July 2024 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
TWTianrui Wang
Tianrui Wang
LZLong Zhou
Long Zhou
ZZZiqiang Zhang
Ziqiang Zhang
YWYu Wu
Yu Wu
SLShujie Liu
Shujie Liu
FWFuru Wei
Furu Wei
Recent research shows a big convergence in model architecture, training objectives, and inference methods across various tasks for different modalities. In this paper, we propose VIOLA, a single auto-regressive Transformer decoder-only network that unifies various cross-modal tasks involving speech and text, such as speech-to-text, text-to-text, text-to-speech, and speech-to-speech tasks, as a conditional language model task via multi-task learning framework. To accomplish this, we first convert the speech utterances to discrete tokens (similar to the textual data) using an offline neural codec encoder. In such a way, all these tasks are converted to token-based sequence prediction problems, which can be naturally handled with one conditional language model. We further integrate task IDs (TID), language IDs (LID), and LSTM-based acoustic embedding into the proposed model to enhance the modeling capability of handling different languages and tasks. Experimental results demonstrate that the proposed VIOLA model can support both single-modal and cross-modal tasks well, and the decoder-only model achieves a comparable and even better performance than the strong baselines. Audio samples are available at https://violademo.github.io/.

23 July 2022 in Circuits, Systems, and Signal Processing
Verified
Journal Article
Closed access
PDF file
ZZZi-Qiang Zhang
Zi-Qiang Zhang
Yan SongYan Song
Yan Song
MWMing-Hui Wu
Ming-Hui Wu
XFXin Fang
Xin Fang
Ian McLoughlinIan McLoughlin
Ian McLoughlin
LDLi-Rong Dai
Li-Rong Dai