Metadata-Version: 2.1
Name: VocalForge
Version: 0.1.1
Summary: Your one-stop solution for voice dataset creation
Author-email: rio@rioharper.com
Classifier: License :: OSI Approved :: MIT License
Classifier: Environment :: GPU :: NVIDIA CUDA :: 11.3
Requires-Python: >=3.8
Description-Content-Type: text/markdown
Provides-Extra: audio
Provides-Extra: text
License-File: LICENSE.md

# An End-to-End Toolkit for Voice Datasets

`VocalForge` is an open-source toolkit written in Python 🐍  that is meant to cut down the time to create datasets for, TTS models, hotword detection models, and more so you can spend more time training, and less time sifting through audio data.

Using [Nvidia's NEMO](https://github.com/NVIDIA/NeMo), [PyAnnote](https://github.com/pyannote/pyannote-audio), [CTC segmentation](https://github.com/lumaku/ctc-segmentation) , [OpenAI's Whisper](https://github.com/openai/whisper), this repo will take you from raw audio to a fully formatted dataset, refining both the audio and text automatically.

*NOTE: While this does reduce time on spent on dataset curation, verifying the output at each step is important as it isn't perfect*

*this is a very much an experimental release, so bugs and updates will be frequent*

![a flow chart of how this repo works](https://github.com/rioharper/VocalForge/blob/main/media/join_processes.svg?raw=true)


## Features:

#### `audio_demo.ipynb`
- ⬇️ **Download audio**  from a YouTube playlist (perfect for podcasts/interviews) OR input your own raw audio files (wav format)
- 🎵 **Remove Non Speech Data**
- 🗣🗣 **Remove Overlapping Speech** 
- 👥 **Split Audio File Into Speakers** 
- 👤 **Isolate the same speaker across multiple files (voice verification)** 
- 🧽 **Use DeepFilterNet to reduce background noise**
- 🧮 **Normalize Audio**
- ➡️ **Export with user defined parameters**

#### `text_demo.ipynb`
- 📜 **Batch transcribe text using OpenAI's Whisper**
- 🧮 **Run [text normalization](https://docs.nvidia.com/deeplearning/nemo/user-guide/docs/en/stable/nlp/text_normalization/wfst/wfst_text_normalization.html)**
- 🫶 **Use CTC segmentation to line up text to audio**
- 🖖 **Split audio based on quality of CTC segmentation confidence**
- ✅ **Generate a metadata.csv and dataset in the format of LJSpeech** 


## Setup/Requirements

Python 3.8 has been tested, newer versions should work

CUDA is required to run all models

a [Hugging Face account](https://huggingface.co/) is required (it's free and super helpful!)

```bash
#install system libraries
apt-get update && apt-get install -y libsndfile1 ffmpeg

conda create -n VocalForge python=3.8 pytorch=1.11.0 torchvision=0.12.0 torchaudio=0.11.0 cudatoolkit=11.3.1 -c pytorch

conda activate VocalForge
#to install from pip
pip install VocalForge
#to install source
git clone https://github.com/rioharper/VocalForge
cd VocalForge
pip install -r requirements.txt

#enter huggingface token, token can be found at https://huggingface.co/settings/tokens
huggingface-cli login
```


Pyannote models need to be "signed up for" in Hugging Face for research purposes. Don't worry, all it asks for is your purpose, website and organization. The following models will have to be manually visited and given the appropriate info:
![an example of signing up for a model](https://github.com/rioharper/VocalForge/blob/main/media/huggingface.png?raw=true)
- [VAD model](https://huggingface.co/pyannote/voice-activity-detection)
- [Overlapped Speech Detection](https://huggingface.co/pyannote/overlapped-speech-detection)
- [Speaker Diarization](https://huggingface.co/pyannote/speaker-diarization)
- [Embedding](https://huggingface.co/pyannote/embedding)
- [Segmentation](https://huggingface.co/pyannote/segmentation)

## API Example
```
from VocalForge.audio import RefineAudio

refine = RefineAudio(
	input_dir='raw_audio', 
	vad_dir='vad', 
	vad_theshold=0.9
)
refine.VoiceDetection.run()
```

## TODO
- [X] Refactor functions for API and toolkit support
- [X] "Sync" datasets with the metadata file if audio clips are deleted after being generated
- [ ] Add a step in the audio refinement processs to remove emotional speech (in progresss)
- [ ] Create a model to remove non-speech utterences and portions with background music (in progresss)
- [X] Update code documentation
- [ ] Add other normalization methods for audio
- [ ] Add other dataset formats for generation
- [ ] Utilize TTS models to automatically generate datasets, with audio augmentation to create diversity
- [ ] Create a Google Colab Notebook
