Model reference · open weights

fastspeech2_conformer

Available as managed deployment Audio espnet Music / audio 1 variants 10k dl/mo

fastspeech2_conformer is an open-weight audio or speech model from espnet. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.

Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.

What it is

Makerespnet
TypeAudio & music
TaskMusic / audio
Runs withtransformers
Released2023-06-06
Popularity10k downloads / month
LicenceOpen weights

About

What fastspeech2_conformer is

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

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]
  • License: Apache 2.0
  • Finetuned from model [optional]: [More Information Needed]

Model Sources [optional]

🤗 Transformers Usage

You can run FastSpeech2Conformer locally with the 🤗 Transformers library.

  1. First install the 🤗 Transformers library, g2p-en:
pip install --upgrade pip
pip install --upgrade transformers g2p-en
  1. Run inference via the Transformers modelling code with the model and hifigan separately

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)
  1. Run inference via the Transformers modelling code with the model and hifigan combined
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)
  1. Run inference with a pipeline and specify which vocoder to use
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

[More Information Needed]

Downstream Use [optional]

[More Information Needed]

Out-of-Scope Use

[More Information Needed]

Bias, Risks, and Limitations

[More Information Needed]

Recommendations

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

[More Information Needed]

Training Procedure

Preprocessing [optional]

[More Information Needed]

Training Hyperparameters

  • Training regime: [More Information Needed]

Speeds, Sizes, Times [optional]

[More Information Needed]

Evaluation

Testing Data, Factors & Metrics

Testing Data

[More Information Needed]

Factors

[More Information Needed]

Metrics

[More Information Needed]

Results

[More Information Needed]

Summary

Model Examination [optional]

[More Information Needed]

Environmental Impact

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]

From the published model card. Full card on the HuggingFace links in the sidebar.

Using it via the API

Call it like any OpenAI endpoint

Once AxForge deploys fastspeech2-conformer for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (fastspeech2-conformer below is illustrative; you get the exact model name on deployment.)

$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \
  -H "Authorization: Bearer $AXFORGE_API_KEY" \
  -F model="fastspeech2-conformer" -F file=@audio.mp3

Create an account — your API key is available in the console. 5M tokens/month currently included with every new account at launch.

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