Preparing Audio Data
The Data tab turns long recordings of one speaker into a training dataset for fine-tuning. Silero VAD detects the speech, Whisper transcribes it, and the result is cut into clips of a few seconds each.
Audio requirements
Works well:
- A single speaker
- Clean recordings with minimal background noise and no music
- 30+ minutes in total; 1-3 hours is better
- Variety of content and speaking styles
Doesn't work:
- Overlapping speakers
- Music beds, heavy noise or reverb
- Very quiet or distorted audio
Quality matters more than quantity: 30 minutes of clean audio beats 3 hours of noisy audio.
Building a dataset
- Put your recordings in one folder. Subfolders are searched too, and files can be any length.Supported formats: WAV, MP3, FLAC, M4A, OGG and Opus.
/path/to/recordings/ ├── episode1.mp3 ├── episode2.wav └── more/chapter3.flac - Open the Data tab.
- Recordings folder: the full path to that folder, e.g.
/home/me/recordingsorC:\Users\me\recordings. - Dataset name: e.g.
my_voice(letters, numbers and underscores). - Whisper model:
openai/whisper-large-v3-turbo(the default) is fast and accurate. Usewhisper-large-v3for the best accuracy, or a smaller model on slow machines. - Language (optional): e.g.
english. If you leave it empty, the language is detected automatically. - Segmentation (optional): Min clip length (default 7 s) and Max clip length (default 20 s).
- Click Build dataset.
When it's done, the dataset appears under Datasets with its number of clips.
Output format
Datasets are saved in LJSpeech format in data/<name>/:
data/my_voice/
├── wavs/
│ ├── my_voice_000000.wav
│ ├── my_voice_000001.wav
│ └── ...
├── metadata.csv
└── info.jsonmetadata.csv has one line per clip, in the form filename|text|text:
my_voice_000000.wav|This is the transcribed text.|This is the transcribed text.You can edit metadata.csv by hand to fix transcription mistakes before training.
Checking quality
Listen to a few random clips in wavs/ and compare them with metadata.csv. Check that:
- each clip starts and ends cleanly, without cut-off words;
- the transcripts match the audio;
- there's no music or second speaker.
If many transcripts are wrong, rebuild with a larger Whisper model or set the language explicitly.
Troubleshooting
- "No speech found in the input files": the audio is too quiet or noisy, or isn't speech. Normalise the volume or use cleaner files.
- Slow: Whisper is slow on CPU. Use a GPU or a smaller Whisper model. You only need to build a dataset once.
- Out of memory: close other GPU apps, or use a smaller Whisper model.
Next step
👉 Finetuning: train a model on your dataset
Copyright and consent
Only train on audio you have the rights to, and only on voices of people who agreed to it.