Hard-Synth: Synthesizing Diverse Hard Samples for ASR using Zero-Shot TTS and LLM
- URL: http://arxiv.org/abs/2411.13159v1
- Date: Wed, 20 Nov 2024 09:49:37 GMT
- Title: Hard-Synth: Synthesizing Diverse Hard Samples for ASR using Zero-Shot TTS and LLM
- Authors: Jiawei Yu, Yuang Li, Xiaosong Qiao, Huan Zhao, Xiaofeng Zhao, Wei Tang, Min Zhang, Hao Yang, Jinsong Su,
- Abstract summary: Text-to-speech (TTS) models have been widely adopted to enhance automatic speech recognition (ASR) systems.
We propose Hard- Synth, a novel ASR data augmentation method that leverages large language models (LLMs) and advanced zero-shot TTS.
Our approach employs LLMs to generate diverse in-domain text through rewriting, without relying on additional text data.
- Score: 48.71951982716363
- License:
- Abstract: Text-to-speech (TTS) models have been widely adopted to enhance automatic speech recognition (ASR) systems using text-only corpora, thereby reducing the cost of labeling real speech data. Existing research primarily utilizes additional text data and predefined speech styles supported by TTS models. In this paper, we propose Hard-Synth, a novel ASR data augmentation method that leverages large language models (LLMs) and advanced zero-shot TTS. Our approach employs LLMs to generate diverse in-domain text through rewriting, without relying on additional text data. Rather than using predefined speech styles, we introduce a hard prompt selection method with zero-shot TTS to clone speech styles that the ASR model finds challenging to recognize. Experiments demonstrate that Hard-Synth significantly enhances the Conformer model, achieving relative word error rate (WER) reductions of 6.5\%/4.4\% on LibriSpeech dev/test-other subsets. Additionally, we show that Hard-Synth is data-efficient and capable of reducing bias in ASR.
Related papers
- MultiVerse: Efficient and Expressive Zero-Shot Multi-Task Text-to-Speech [7.038489351956803]
MultiVerse is a zero-shot multi-task TTS system that is able to perform TTS or speech style transfer in zero-shot and cross-lingual conditions.
We use source-filter theory-based disentanglement, utilizing the prompt for modeling filter-related and source-related representations.
Our novel prosody modeling technique significantly contributes to MultiVerse's ability to generate speech with high prosody similarity to the given prompts.
arXiv Detail & Related papers (2024-10-04T07:10:25Z) - Text-To-Speech Synthesis In The Wild [76.71096751337888]
Text-to-speech (TTS) systems are traditionally trained using modest databases of studio-quality, prompted or read speech collected in benign acoustic environments such as anechoic rooms.
We introduce the TTS In the Wild (TITW) dataset, the result of a fully automated pipeline, applied to the VoxCeleb1 dataset commonly used for speaker recognition.
We show that a number of recent TTS models can be trained successfully using TITW-Easy, but that it remains extremely challenging to produce similar results using TITW-Hard.
arXiv Detail & Related papers (2024-09-13T10:58:55Z) - SimpleSpeech 2: Towards Simple and Efficient Text-to-Speech with Flow-based Scalar Latent Transformer Diffusion Models [64.40250409933752]
We build upon our previous publication by implementing a simple and efficient non-autoregressive (NAR) TTS framework, termed SimpleSpeech 2.
SimpleSpeech 2 effectively combines the strengths of both autoregressive (AR) and non-autoregressive (NAR) methods.
We show a significant improvement in generation performance and generation speed compared to our previous work and other state-of-the-art (SOTA) large-scale TTS models.
arXiv Detail & Related papers (2024-08-25T17:07:39Z) - On the Problem of Text-To-Speech Model Selection for Synthetic Data Generation in Automatic Speech Recognition [31.58289343561422]
We compare five different TTS decoder architectures in the scope of synthetic data generation to show the impact on CTC-based speech recognition training.
For data generation auto-regressive decoding performs better than non-autoregressive decoding, and propose an approach to quantify TTS generalization capabilities.
arXiv Detail & Related papers (2024-07-31T09:37:27Z) - TextrolSpeech: A Text Style Control Speech Corpus With Codec Language
Text-to-Speech Models [51.529485094900934]
We propose TextrolSpeech, which is the first large-scale speech emotion dataset annotated with rich text attributes.
We introduce a multi-stage prompt programming approach that effectively utilizes the GPT model for generating natural style descriptions in large volumes.
To address the need for generating audio with greater style diversity, we propose an efficient architecture called Salle.
arXiv Detail & Related papers (2023-08-28T09:06:32Z) - Towards Selection of Text-to-speech Data to Augment ASR Training [20.115236045164355]
We train a neural network to measure the similarity of a synthetic data to real speech.
We find that incorporating synthetic samples with considerable dissimilarity to real speech is crucial for boosting recognition performance.
arXiv Detail & Related papers (2023-05-30T17:24:28Z) - A Vector Quantized Approach for Text to Speech Synthesis on Real-World
Spontaneous Speech [94.64927912924087]
We train TTS systems using real-world speech from YouTube and podcasts.
Recent Text-to-Speech architecture is designed for multiple code generation and monotonic alignment.
We show thatRecent Text-to-Speech architecture outperforms existing TTS systems in several objective and subjective measures.
arXiv Detail & Related papers (2023-02-08T17:34:32Z) - Distribution augmentation for low-resource expressive text-to-speech [18.553812159109253]
This paper presents a novel data augmentation technique for text-to-speech (TTS)
It allows to generate new (text, audio) training examples without requiring any additional data.
arXiv Detail & Related papers (2022-02-13T21:19:31Z) - STYLER: Style Modeling with Rapidity and Robustness via
SpeechDecomposition for Expressive and Controllable Neural Text to Speech [2.622482339911829]
STYLER is a novel expressive text-to-speech model with parallelized architecture.
Our novel noise modeling approach from audio using domain adversarial training and Residual Decoding enabled style transfer without transferring noise.
arXiv Detail & Related papers (2021-03-17T07:11:09Z) - 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)
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.