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In speech separation, time-domain approaches have successfully replaced the time-frequency domain with latent sequence feature from a learnable encoder. Conventionally, the feature is separated into speaker-specific ones at the final stage…

音频与语音处理 · 电气工程与系统科学 2026-04-01 Ui-Hyeop Shin , Sangyoun Lee , Taehan Kim , Hyung-Min Park

We present an extension to the Tacotron speech synthesis architecture that learns a latent embedding space of prosody, derived from a reference acoustic representation containing the desired prosody. We show that conditioning Tacotron on…

计算与语言 · 计算机科学 2018-03-28 RJ Skerry-Ryan , Eric Battenberg , Ying Xiao , Yuxuan Wang , Daisy Stanton , Joel Shor , Ron J. Weiss , Rob Clark , Rif A. Saurous

Non-parallel voice conversion aims to convert voice from a source domain to a target domain without paired training data. Cycle-Consistent Generative Adversarial Networks (CycleGAN) and Variational Autoencoders (VAE) have been used for this…

声音 · 计算机科学 2025-10-16 Maharnab Saikia

By representing speaker characteristic as a single fixed-length vector extracted solely from speech, we can train a neural multi-speaker speech synthesis model by conditioning the model on those vectors. This model can also be adapted to…

音频与语音处理 · 电气工程与系统科学 2019-10-09 Hieu-Thi Luong , Junichi Yamagishi

We propose a neural network for zero-shot voice conversion (VC) without any parallel or transcribed data. Our approach uses pre-trained models for automatic speech recognition (ASR) and speaker embedding, obtained from a speaker…

音频与语音处理 · 电气工程与系统科学 2020-05-19 Yurii Rebryk , Stanislav Beliaev

Large speech generation models are evolving from single-speaker, short sentence synthesis to multi-speaker, long conversation geneartion. Current long-form speech generation models are predominately constrained to dyadic, turn-based…

Existing singing voice synthesis models (SVS) are usually trained on singing data and depend on either error-prone time-alignment and duration features or explicit music score information. In this paper, we propose Karaoker, a multispeaker…

Nowadays vast amounts of speech data are recorded from low-quality recorder devices such as smartphones, tablets, laptops, and medium-quality microphones. The objective of this research was to study the automatic generation of high-quality…

Speech-to-face generation is an intriguing area of research that focuses on generating realistic facial images based on a speaker's audio speech. However, state-of-the-art methods employing GAN-based architectures lack stability and cannot…

计算机视觉与模式识别 · 计算机科学 2023-10-06 Jinting Wang , Li Liu , Jun Wang , Hei Victor Cheng

Speaker-role diarization (RD), such as doctor vs. patient or lawyer vs. client, is practically often more useful than conventional speaker diarization (SD), which assigns only generic labels (speaker-1, speaker-2). The state-of-the-art…

音频与语音处理 · 电气工程与系统科学 2025-12-23 Arindam Ghosh , Mark Fuhs , Bongjun Kim , Anurag Chowdhury , Monika Woszczyna

The effects of speaking-style variability on automatic speaker verification were investigated using the UCLA Speaker Variability database which comprises multiple speaking styles per speaker. An x-vector/PLDA (probabilistic linear…

音频与语音处理 · 电气工程与系统科学 2020-08-11 Amber Afshan , Jinxi Guo , Soo Jin Park , Vijay Ravi , Alan McCree , Abeer Alwan

Recently, mainstream mel-spectrogram-based neural vocoders rely on generative adversarial network (GAN) for high-fidelity speech generation, e.g., HiFi-GAN and BigVGAN. However, the use of GAN restricts training efficiency and model…

音频与语音处理 · 电气工程与系统科学 2025-08-12 Hui-Peng Du , Yang Ai , Rui-Chen Zheng , Ye-Xin Lu , Zhen-Hua Ling

The current trend in automatic speech recognition is to leverage large amounts of labeled data to train supervised neural network models. Unfortunately, obtaining data for a wide range of domains to train robust models can be costly.…

计算与语言 · 计算机科学 2018-06-14 Wei-Ning Hsu , Hao Tang , James Glass

Expressive speech synthesis requires vibrant prosody and well-timed pauses. We propose an effective strategy to augment a small dataset to train an expressive end-to-end Text-to-Speech model. We merge audios of emotionally congruent text…

声音 · 计算机科学 2026-02-12 Raymond Chung

In recent years, several text-to-speech systems have been proposed to synthesize natural speech in zero-shot, few-shot, and low-resource scenarios. However, these methods typically require training with data from many different speakers.…

音频与语音处理 · 电气工程与系统科学 2025-06-03 Kishor Kayyar Lakshminarayana , Frank Zalkow , Christian Dittmar , Nicola Pia , Emanuel A. P. Habets

Modern approaches to text to speech require the entire input character sequence to be processed before any audio is synthesised. This latency limits the suitability of such models for time-sensitive tasks like simultaneous interpretation.…

音频与语音处理 · 电气工程与系统科学 2021-06-16 Devang S Ram Mohan , Raphael Lenain , Lorenzo Foglianti , Tian Huey Teh , Marlene Staib , Alexandra Torresquintero , Jiameng Gao

Neural audio codecs are at the core of modern conversational speech technologies, converting continuous speech into sequences of discrete tokens that can be processed by LLMs. However, existing codecs typically operate at fixed frame rates,…

机器学习 · 计算机科学 2026-02-05 Luca Della Libera , Cem Subakan , Mirco Ravanelli

In this paper, we propose a model to perform style transfer of speech to singing voice. Contrary to the previous signal processing-based methods, which require high-quality singing templates or phoneme synchronization, we explore a…

声音 · 计算机科学 2022-08-29 Shrutina Agarwal , Sriram Ganapathy , Naoya Takahashi

Sequence-to-sequence models provide a simple and elegant solution for building speech recognition systems by folding separate components of a typical system, namely acoustic (AM), pronunciation (PM) and language (LM) models into a single…

音频与语音处理 · 电气工程与系统科学 2017-12-06 Bo Li , Tara N. Sainath , Khe Chai Sim , Michiel Bacchiani , Eugene Weinstein , Patrick Nguyen , Zhifeng Chen , Yonghui Wu , Kanishka Rao

Sequence-to-sequence (seq2seq) models are competitive with hybrid models for automatic speech recognition (ASR) tasks when large amounts of training data are available. However, data sparsity and domain adaptation are more problematic for…

计算与语言 · 计算机科学 2021-06-16 Chak-Fai Li , Francis Keith , William Hartmann , Matthew Snover , Owen Kimball