Datasets:
raw-emocean
Large-scale English speech dataset for text-to-speech (TTS) model training. Designed for autoregressive TTS architectures (TADA, CSM, VALL-E style models).
Dataset Summary
| Metric | Value |
|---|---|
| Parquet shards | 7 |
| Segment duration | 3–8 seconds |
| Sample rate | 24,000 Hz (mono) |
| ASR engine | NVIDIA Parakeet TDT 0.6B v3 |
| Format | Parquet with embedded audio |
Dataset Schema
| Column | Type | Description |
|---|---|---|
audio |
Audio |
Waveform array + sampling rate (24kHz mono) |
text |
string |
ASR transcript (Parakeet TDT 0.6B v3, no normalization) |
duration |
float64 |
Segment duration in seconds |
snr |
float64 |
Estimated signal-to-noise ratio (dB) |
video_id |
string |
YouTube video identifier |
source_url |
string |
Full source URL |
start_time |
float64 |
Segment start offset in source audio (seconds) |
end_time |
float64 |
Segment end offset in source audio (seconds) |
Quick Start
from datasets import load_dataset
# Stream large dataset without downloading entirely
ds = load_dataset("somu9/raw-emocean", split="train", streaming=True)
for sample in ds:
print(sample["text"])
break
# Load full dataset
ds = load_dataset("somu9/raw-emocean", split="train")
sample = ds[0]
audio_array = sample["audio"]["array"] # numpy array
sample_rate = sample["audio"]["sampling_rate"] # 24000
transcript = sample["text"]
# Filter by duration or SNR
clean = ds.filter(lambda x: x["snr"] > 30 and x["duration"] > 4)
Data Collection Pipeline
YouTube Audio
│
▼
yt-dlp download (WAV 24kHz)
│
▼
Voice Activity Detection (Silero VAD)
│ threshold=0.5, min_speech=500ms, min_silence=300ms
▼
Segment Extraction (3–8s segments)
│
▼
Quality Filters
│ ├─ SNR > 25 dB
│ ├─ Clipping < 0.1%
│ ├─ Speaker overlap detection (pitch bimodality)
│ └─ Music detection (spectral flatness + flux)
│
▼
ASR Transcription (Parakeet TDT 0.6B v3)
│ batch_size=128, CUDA
▼
manifest.csv + audio/*.wav
Quality Assurance
- SNR filtering: Segments below 25 dB signal-to-noise ratio are discarded
- Clipping detection: Segments with >0.1% of samples at peak amplitude are removed
- Speaker overlap: Pitch-based bimodality detection removes segments with simultaneous speakers
- Music detection: Spectral flatness and flux analysis removes segments with background music
- Minimum length: Segments shorter than 3 seconds are discarded
- Empty transcript filter: Segments where ASR produces fewer than 5 characters are removed
Intended Use
- Pre-training autoregressive TTS models (TADA, CSM, VALL-E, SoundStorm)
- Fine-tuning speech synthesis models
- Speech representation learning
- ASR training data augmentation
Limitations
- Transcripts are ASR-generated (Parakeet) — not human-verified, expect ~5% error rate
- Audio sourced from YouTube — variable recording conditions across sources
- No speaker identity labels — segments are not diarized
- No emotion or prosody annotations
- English only
Citation
If you use this dataset, please cite:
@dataset{raw-emocean_2026,
title = {raw-emocean: English Speech Dataset for TTS Training},
author = {Dataset Contributors},
year = {2026},
url = {https://huggingface.co/datasets/somu9/raw-emocean}
}
License
CC-BY-4.0
Last updated: 2026-04-25 14:27 UTC
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