TTS-Transducer: End-to-End Speech Synthesis with Neural Transducer
- URL: http://arxiv.org/abs/2501.06320v1
- Date: Fri, 10 Jan 2025 19:50:32 GMT
- Title: TTS-Transducer: End-to-End Speech Synthesis with Neural Transducer
- Authors: Vladimir Bataev, Subhankar Ghosh, Vitaly Lavrukhin, Jason Li,
- Abstract summary: TTS-Transducer is a novel architecture for text-to-speech, leveraging the strengths of audio models and neural transducers.
We show that TTS-Transducer is a competitive and robust alternative to contemporary TTS systems.
- Score: 6.1319363847980135
- License:
- Abstract: This work introduces TTS-Transducer - a novel architecture for text-to-speech, leveraging the strengths of audio codec models and neural transducers. Transducers, renowned for their superior quality and robustness in speech recognition, are employed to learn monotonic alignments and allow for avoiding using explicit duration predictors. Neural audio codecs efficiently compress audio into discrete codes, revealing the possibility of applying text modeling approaches to speech generation. However, the complexity of predicting multiple tokens per frame from several codebooks, as necessitated by audio codec models with residual quantizers, poses a significant challenge. The proposed system first uses a transducer architecture to learn monotonic alignments between tokenized text and speech codec tokens for the first codebook. Next, a non-autoregressive Transformer predicts the remaining codes using the alignment extracted from transducer loss. The proposed system is trained end-to-end. We show that TTS-Transducer is a competitive and robust alternative to contemporary TTS systems.
Related papers
- Alignment-Free Training for Transducer-based Multi-Talker ASR [55.1234384771616]
Multi-talker RNNT (MT-RNNT) aims to achieve recognition without relying on costly front-end source separation.
We propose a novel alignment-free training scheme for the MT-RNNT (MT-RNNT-AFT) that adopts the standard RNNT architecture.
arXiv Detail & Related papers (2024-09-30T13:58:11Z) - CosyVoice: A Scalable Multilingual Zero-shot Text-to-speech Synthesizer based on Supervised Semantic Tokens [49.569695524535454]
We propose to represent speech with supervised semantic tokens, which are derived from a multilingual speech recognition model by inserting vector quantization into the encoder.
Based on the tokens, we further propose a scalable zero-shot TTS synthesizer, CosyVoice, which consists of an LLM for text-to-token generation and a conditional flow matching model for token-to-speech synthesis.
arXiv Detail & Related papers (2024-07-07T15:16:19Z) - Improving Robustness of LLM-based Speech Synthesis by Learning Monotonic Alignment [19.48653924804823]
Large Language Model (LLM) based text-to-speech (TTS) systems have demonstrated remarkable capabilities in handling large speech datasets and generating natural speech for new speakers.
However, LLM-based TTS models are not robust as the generated output can contain repeating words, missing words and mis-aligned speech.
We examine these challenges in an encoder-decoder transformer model and find that certain cross-attention heads in such models implicitly learn the text and speech alignment when trained for predicting speech tokens for a given text.
arXiv Detail & Related papers (2024-06-25T22:18:52Z) - TVLT: Textless Vision-Language Transformer [89.31422264408002]
We present the Textless Vision-Language Transformer (TVLT), where homogeneous transformer blocks take raw visual and audio inputs.
TVLT attains performance comparable to its text-based counterpart, on various multimodal tasks.
Our findings suggest the possibility of learning compact and efficient visual-linguistic representations from low-level visual and audio signals.
arXiv Detail & Related papers (2022-09-28T15:08:03Z) - Pre-Training Transformer Decoder for End-to-End ASR Model with Unpaired
Speech Data [145.95460945321253]
We introduce two pre-training tasks for the encoder-decoder network using acoustic units, i.e., pseudo codes.
The proposed Speech2C can relatively reduce the word error rate (WER) by 19.2% over the method without decoder pre-training.
arXiv Detail & Related papers (2022-03-31T15:33:56Z) - Advances in Speech Vocoding for Text-to-Speech with Continuous
Parameters [2.6572330982240935]
This paper presents new techniques in a continuous vocoder, that is all features are continuous and presents a flexible speech synthesis system.
New continuous noise masking based on the phase distortion is proposed to eliminate the perceptual impact of the residual noise.
Bidirectional long short-term memory (LSTM) and gated recurrent unit (GRU) are studied and applied to model continuous parameters for more natural-sounding like a human.
arXiv Detail & Related papers (2021-06-19T12:05:01Z) - Stacked Acoustic-and-Textual Encoding: Integrating the Pre-trained
Models into Speech Translation Encoders [30.160261563657947]
Speech-to-translation data is scarce; pre-training is promising in end-to-end Speech Translation.
We propose a Stacked.
Acoustic-and-Textual (SATE) method for speech translation.
Our encoder begins with processing the acoustic sequence as usual, but later behaves more like an.
MT encoder for a global representation of the input sequence.
arXiv Detail & Related papers (2021-05-12T16:09:53Z) - GraphSpeech: Syntax-Aware Graph Attention Network For Neural Speech
Synthesis [79.1885389845874]
Transformer-based end-to-end text-to-speech synthesis (TTS) is one of such successful implementations.
We propose a novel neural TTS model, denoted as GraphSpeech, that is formulated under graph neural network framework.
Experiments show that GraphSpeech consistently outperforms the Transformer TTS baseline in terms of spectrum and prosody rendering of utterances.
arXiv Detail & Related papers (2020-10-23T14:14:06Z) - MultiSpeech: Multi-Speaker Text to Speech with Transformer [145.56725956639232]
Transformer-based text to speech (TTS) model (e.g., Transformer TTSciteli 2019neural, FastSpeechciteren 2019fastspeech) has shown the advantages of training and inference efficiency over RNN-based model.
We develop a robust and high-quality multi-speaker Transformer TTS system called MultiSpeech, with several specially designed components/techniques to improve text-to-speech alignment.
arXiv Detail & Related papers (2020-06-08T15:05:28Z)
This list is automatically generated from the titles and abstracts of the papers in this site.
This site does not guarantee the quality of this site (including all information) and is not responsible for any consequences.