fastspeech2 conformer

Providerespnet
Categorytext-to-audio
Licenseapache-2.0
Downloads9.6K
Stars0

Overview

FastSpeech 2 Conformer is a non-autoregressive text-to-speech (TTS) model designed for high-fidelity audio synthesis with significantly reduced inference latency compared to traditional RNN-based architectures. By integrating Conformer blocks—which combine convolutional layers for local feature extraction and self-attention for global context—the model achieves superior prosody and naturalness while maintaining a fast, parallelizable generation speed. For developers, this means a scalable solution for real-time voice applications, such as virtual assistants or automated narration, without the stability issues common in autoregressive models. It integrates well into pipelines requiring Apache-2.0 licensed components and is optimized for efficiency across various hardware targets.

Highlights

  • Non-autoregressive architecture for ultra-fast inference speeds
  • Conformer blocks improve prosody and audio naturalness
  • Apache-2.0 license ensures flexible commercial integration
  • Eliminates common stability issues found in autoregressive TTS
  • Ideal for real-time voice synthesis and scalable deployments

Usage

Install
# Install Hugging Face transformers
pip install transformers torch
SDK Usage
# Load model with transformers
from transformers import AutoModel, AutoTokenizer

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

Hugging Face Download

We recommend downloading the model via the Hugging Face CLI or Hub SDK.

Guidance:Before downloading, install huggingface_hub with:

Guidance
pip install -U huggingface_hub

CLI Download

Download the full repository

Download the full repository
huggingface-cli download espnet/fastspeech2_conformer

Download a single file to a local folder (e.g. config.json into ./dir)

Download a single file to a local folder (e.g. config.json into ./dir)
huggingface-cli download espnet/fastspeech2_conformer config.json --local-dir ./dir

See the official docs for more CLI options

SDK Download

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

Git Download

Make sure git-lfs is installed first

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

To skip LFS large-file downloads, use:

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

Model files are hosted on the Hugging Face Hub — download directly via HF CLI / SDK / Git, not through this site.

PyTorch / Transformers Usage

Install Transformers

Install Transformers
pip install -U transformers torch

Load the model and run inference

Load the model and run inference
from transformers import AutoModelForCausalLM, AutoTokenizer

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

Full Documentation

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]

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🤗 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

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[More Information Needed]

Recommendations

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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

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#### 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]

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[More Information Needed]

Evaluation

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

Testing Data, Factors & Metrics

#### Testing Data

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[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

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#### 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

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