fastspeech2 conformer

提供商espnet
分类text-to-audio
许可证apache-2.0
下载量9.6K
星标0

简介

FastSpeech2 Conformer 是由 ESPnet 提供的文本转语音(TTS)模型,它在 FastSpeech2 的非自回归架构基础上引入了 Conformer 模块,通过结合卷积和自注意力机制,显著提升了对语音韵律和细节的建模能力。相比传统 TTS,它在保证推理速度(快于实时)的同时,解决了语音生硬的问题,合成音质更自然。对于开发者而言,该模型适合用于构建高性能的语音助手或自动化配音系统,上手难度中等,通常作为语音合成管线中的声学模型,需配合 Vocoder(如 HiFi-GAN)将频谱图转换为最终音频。

核心亮点

  • 结合 Conformer 架构,语音韵律更自然
  • 非自回归推理,合成速度极快且稳定
  • 适配 ESPnet 框架,适合工业级 TTS 部署
  • Apache-2.0 协议,对商业应用非常友好

使用方法

安装依赖
# 安装 Hugging Face transformers
pip install transformers torch
SDK 使用
# 使用 transformers 加载模型
from transformers import AutoModel, AutoTokenizer

model = AutoModel.from_pretrained("espnet/fastspeech2_conformer")
tokenizer = AutoTokenizer.from_pretrained("espnet/fastspeech2_conformer")

Hugging Face 下载

我们推荐使用命令行或者 Hugging Face Hub SDK 来进行模型的下载。

操作指引:在下载前,请先通过如下命令安装 huggingface_hub:

操作指引
pip install -U huggingface_hub

命令行下载

下载完整模型库

下载完整模型库
huggingface-cli download espnet/fastspeech2_conformer

下载单个文件到指定本地文件夹(以下载 config.json 到当前路径下 ./dir 目录为例)

下载单个文件到指定本地文件夹(以下载 config.json 到当前路径下 ./dir 目录为例)
huggingface-cli download espnet/fastspeech2_conformer config.json --local-dir ./dir

更多命令行下载选项,可参见官方文档

SDK 下载

SDK 下载
# 模型下载
from huggingface_hub import snapshot_download
model_dir = snapshot_download('espnet/fastspeech2_conformer')

Git 下载

请确保 lfs 已经被正确安装

Git 下载
git lfs install
git clone https://huggingface.co/espnet/fastspeech2_conformer

如果您希望跳过 lfs 大文件下载,可以使用如下命令

跳过 LFS
GIT_LFS_SKIP_SMUDGE=1 git clone https://huggingface.co/espnet/fastspeech2_conformer

模型文件托管在 Hugging Face Hub,使用 HF CLI / SDK / Git 直接下载,不经过本站。

PyTorch / Transformers 使用

安装 Transformers

安装 Transformers
pip install -U transformers torch

模型加载和推理

模型加载和推理
from transformers import AutoModelForCausalLM, AutoTokenizer

model = AutoModelForCausalLM.from_pretrained('espnet/fastspeech2_conformer')
tokenizer = AutoTokenizer.from_pretrained('espnet/fastspeech2_conformer')

完整文档

来源: HuggingFace

---
license: apache-2.0
language:

  • en

library_name: transformers
pipeline_tag: text-to-audio
tags:
  • audio

---

FastSpeech2Conformer

<!-- Provide a quick summary of what the model is/does. -->

FastSpeech2Conformer is a non-autoregressive text-to-speech (TTS) model that combines the strengths of FastSpeech2 and the conformer architecture to generate high-quality speech from text quickly and efficiently.

Model Description

<!-- Provide a longer summary of what this model is. -->

The FastSpeech2Conformer model was proposed with the paper Recent Developments On Espnet Toolkit Boosted By Conformer by Pengcheng Guo, Florian Boyer, Xuankai Chang, Tomoki Hayashi, Yosuke Higuchi, Hirofumi Inaguma, Naoyuki Kamo, Chenda Li, Daniel Garcia-Romero, Jiatong Shi, Jing Shi, Shinji Watanabe, Kun Wei, Wangyou Zhang, and Yuekai Zhang. It was first released in this repository. The license used is Apache 2.0.

FastSpeech2 is a non-autoregressive TTS model, which means it can generate speech significantly faster than autoregressive models. It addresses some of the limitations of its predecessor, FastSpeech, by directly training the model with ground-truth targets instead of the simplified output from a teacher model. It also introduces more variation information of speech (e.g., pitch, energy, and more accurate duration) as conditional inputs. Furthermore, the conformer (convolutional transformer) architecture makes use of convolutions inside the transformer blocks to capture local speech patterns, while the attention layer is able to capture relationships in the input that are farther away.

  • Developed by: Pengcheng Guo, Florian Boyer, Xuankai Chang, Tomoki Hayashi, Yosuke Higuchi, Hirofumi Inaguma, Naoyuki Kamo, Chenda Li, Daniel Garcia-Romero, Jiatong Shi, Jing Shi, Shinji Watanabe, Kun Wei, Wangyou Zhang, and Yuekai Zhang.
  • Shared by: Connor Henderson
  • Model type: text-to-speech
  • Language(s) (NLP): [More Information Needed]
  • Finetuned from model [optional]: [More Information Needed]

Model Sources [optional]

<!-- Provide the basic links for the model. -->

🤗 Transformers Usage

You can run FastSpeech2Conformer locally with the 🤗 Transformers library.

1. First install the 🤗 Transformers library, g2p-en:

code
pip install --upgrade pip
pip install --upgrade transformers g2p-en

2. Run inference via the Transformers modelling code with the model and hifigan separately

python
from transformers import FastSpeech2ConformerTokenizer, FastSpeech2ConformerModel, FastSpeech2ConformerHifiGan
import soundfile as sf

tokenizer = FastSpeech2ConformerTokenizer.from_pretrained("espnet/fastspeech2_conformer")
inputs = tokenizer("Hello, my dog is cute.", return_tensors="pt")
input_ids = inputs["input_ids"]

model = FastSpeech2ConformerModel.from_pretrained("espnet/fastspeech2_conformer")
output_dict = model(input_ids, return_dict=True)
spectrogram = output_dict["spectrogram"]

hifigan = FastSpeech2ConformerHifiGan.from_pretrained("espnet/fastspeech2_conformer_hifigan")
waveform = hifigan(spectrogram)

sf.write("speech.wav", waveform.squeeze().detach().numpy(), samplerate=22050)

3. Run inference via the Transformers modelling code with the model and hifigan combined

python
from transformers import FastSpeech2ConformerTokenizer, FastSpeech2ConformerWithHifiGan
import soundfile as sf

tokenizer = FastSpeech2ConformerTokenizer.from_pretrained("espnet/fastspeech2_conformer")
inputs = tokenizer("Hello, my dog is cute.", return_tensors="pt")
input_ids = inputs["input_ids"]

model = FastSpeech2ConformerWithHifiGan.from_pretrained("espnet/fastspeech2_conformer_with_hifigan")
output_dict = model(input_ids, return_dict=True)
waveform = output_dict["waveform"]

sf.write("speech.wav", waveform.squeeze().detach().numpy(), samplerate=22050)

4. Run inference with a pipeline and specify which vocoder to use

python
from transformers import pipeline, FastSpeech2ConformerHifiGan
import soundfile as sf

vocoder = FastSpeech2ConformerHifiGan.from_pretrained("espnet/fastspeech2_conformer_hifigan")
synthesiser = pipeline(model="espnet/fastspeech2_conformer", vocoder=vocoder)

speech = synthesiser("Hello, my dog is cooler than you!")

sf.write("speech.wav", speech["audio"].squeeze(), samplerate=speech["sampling_rate"])

Direct Use

<!-- This section is for the model use without fine-tuning or plugging into a larger ecosystem/app. -->

[More Information Needed]

Downstream Use [optional]

<!-- This section is for the model use when fine-tuned for a task, or when plugged into a larger ecosystem/app -->

[More Information Needed]

Out-of-Scope Use

<!-- This section addresses misuse, malicious use, and uses that the model will not work well for. -->

[More Information Needed]

Bias, Risks, and Limitations

<!-- This section is meant to convey both technical and sociotechnical limitations. -->

[More Information Needed]

Recommendations

<!-- This section is meant to convey recommendations with respect to the bias, risk, and technical limitations. -->

Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. More information needed for further recommendations.

How to Get Started with the Model

Use the code below to get started with the model.

[More Information Needed]

Training Details

Training Data

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[More Information Needed]

Training Procedure

<!-- This relates heavily to the Technical Specifications. Content here should link to that section when it is relevant to the training procedure. -->

#### Preprocessing [optional]

[More Information Needed]

#### Training Hyperparameters

  • Training regime: [More Information Needed] <!--fp32, fp16 mixed precision, bf16 mixed precision, bf16 non-mixed precision, fp16 non-mixed precision, fp8 mixed precision -->

#### Speeds, Sizes, Times [optional]

<!-- This section provides information about throughput, start/end time, checkpoint size if relevant, etc. -->

[More Information Needed]

Evaluation

<!-- This section describes the evaluation protocols and provides the results. -->

Testing Data, Factors & Metrics

#### Testing Data

<!-- This should link to a Data Card if possible. -->

[More Information Needed]

#### Factors

<!-- These are the things the evaluation is disaggregating by, e.g., subpopulations or domains. -->

[More Information Needed]

#### Metrics

<!-- These are the evaluation metrics being used, ideally with a description of why. -->

[More Information Needed]

Results

[More Information Needed]

#### Summary

Model Examination [optional]

<!-- Relevant interpretability work for the model goes here -->

[More Information Needed]

Environmental Impact

<!-- Total emissions (in grams of CO2eq) and additional considerations, such as electricity usage, go here. Edit the suggested text below accordingly -->

Carbon emissions can be estimated using the Machine Learning Impact calculator presented in Lacoste et al. (2019).

  • Hardware Type: [More Information Needed]
  • Hours used: [More Information Needed]
  • Cloud Provider: [More Information Needed]
  • Compute Region: [More Information Needed]
  • Carbon Emitted: [More Information Needed]

Technical