Env-TTS-SD-Corpus / README.md
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---
license: cc-by-sa-4.0
language:
- en
- zh
tags:
- env-tts
- environment-aware-tts
- speaker-diarization
- text-to-speech
- audio
- speech-synthesis
size_categories:
- 100K<n<1M
pretty_name: env-tts-sd-corpus
---
# env-tts-sd-corpus
**Environment-aware text-to-speech training corpus.** Each row pairs three short
16 kHz mono FLAC clips with a transcript:
- an **environment** sample (different speaker, same acoustic scene),
- a **speaker** reference (same speaker, optionally with augmented acoustics),
- the target **speech**,
so a TTS model can learn to synthesise a target utterance with both a specified
voice and a specified environment.
## Schema
| column | type | description |
| --- | --- | --- |
| `environment_audio_source` | binary (FLAC 16 kHz mono) | acoustic-scene reference, ≤15 s, drawn from a different speaker than `speech` but the same recording session |
| `environment_audio_duration` | float32 | seconds |
| `speaker_audio_source` | binary (FLAC 16 kHz mono) | speaker-identity reference, ≤15 s, from the same speaker as `speech` |
| `speaker_audio_duration` | float32 | seconds |
| `text` | string | transcript of `speech`; either the original gold transcript or a fresh Qwen3-ASR re-label |
| `speech` | binary (FLAC 16 kHz mono) | target utterance, 3–15 s |
| `speech_duration` | float32 | seconds |
| `language` | string | `zh` / `en` / `auto` |
| `dataset` | string | one of `m3sd` / `aishell4` / `msdwild` / `chime6` |
| `conversation_id` | string | unique within the source dataset |
| `speaker_id` | string | within-conversation diarisation label |
| `env_id` | string | acoustic-scene identifier (usually the conversation_id) |
| `text_source` | string | `original` or `asr` |
| `spk_aug` | string | `none` / `noise` / `rir+noise` *(only present when augmentation was applied)* |
| `spk_aug_snr_db` | float32 | signal-to-noise ratio used when `spk_aug != none` |
## Source corpora
| dataset | hours | sessions | language | how we use it |
| --- | ---: | ---: | --- | --- |
| **M3SD** (Wu et al., 2025) | 770 | 1 372 | zh / en mixed | YouTube speaker-diarisation corpus, multi-scenario, transcripts via Qwen3-ASR |
| **AISHELL-4** (Fu et al., 2021) | 120 | 211 | zh | Mandarin meetings with native TextGrid transcripts |
| **MSDWILD** (Liu et al., 2022) | 80 | 3 143 | zh / en mixed | in-the-wild speaker-diarisation videos, transcripts via Qwen3-ASR |
| **CHiME-6** (Watanabe et al., 2020) | 40 | 18 | en | dinner-party recordings (Kinect U06 / U01 binaural), official JSON transcripts |
## Processing pipeline
The pipeline is three streaming stages running in parallel as separate
processes, with a small filesystem-based handoff for backpressure:
1. **download** — one thread per source. Hugging Face mirrors and direct
tar URLs are streamed with `httpx.stream`; the tar bytes are never written
to disk in full. Each upstream "conversation" emits a JSON sentinel under
`state/ready/` as soon as its audio is locally addressable.
2. **process** — an asyncio loop drains sentinels with a bounded semaphore
(default 64 concurrent conversations). For each conversation it
- resamples the audio to 16 kHz mono,
- walks the diarisation turns, chunks each turn into 3–15 s pieces,
- picks a same-speaker reference (≥3 s, concatenating short turns when
needed) and a different-speaker environment slice (≥3 s, extended into
surrounding audio if necessary),
- submits any speech pieces whose transcript is missing or whose turn was
split mid-utterance to a **Qwen3-ASR-1.7B** Flask worker for fresh ASR,
- FLAC-encodes the three clips and appends a row to the sharded parquet
writer.
The ASR worker coalesces concurrent requests into length-bucketed batches
([0–4 s], [4–9 s], [9–16 s], 16 s+) so that the HuggingFace
`padding=True` step inside `qwen-asr` does not waste GPU on long zero-pad
tails. Single-clip OOMs are dropped silently (the row is dropped, not the
sibling 255 clips).
3. **upload** — watches `final/upload_queue/group_*/` for sealed groups and
uploads them to this repo via `HfApi.upload_folder`. Each group bundles
≈3 200 rows (4 parquet shards × 800 rows). Commits are rate-limited.
The reader, ASR worker, augmenter, and writer are all designed to recover
cleanly from `SIGKILL`: all state is captured in a few small JSON files under
`state/` and an atomic-rename `.tmp → final` write protocol for each parquet
shard.
## Provenance: row breakdown by dataset
Counts are approximate (depend on streaming end + final partial groups).
| dataset | records | rows emitted |
| --- | ---: | ---: |
| M3SD | 1 372 | ≈212 000 |
| MSDWILD | 3 113 | ≈ 28 400 |
| AISHELL-4 | 145 | ≈ 35 250 |
| CHiME-6 | 18 | ≈ 15 800 |
Drop reasons that account for "records ingested" > "records emitted with
rows" in MSDWILD/M3SD: conversations with <2 speakers (no candidate for the
env source), conversations whose total speech time per speaker can't yield
a ≥3 s focal clip + a ≥3 s reference, or clips where `Qwen3-ASR` returned
empty text after segmentation.
## Licensing
The derived corpus is released under **CC-BY-SA-4.0**, which inherits the
most-restrictive licence among the four sources. Note in particular:
- **M3SD** is for *academic and non-commercial research* only (Wu et al., 2025).
- **MSDWILD** uses the X-LANCE research-only agreement (Liu et al., 2022).
- AISHELL-4 (Apache 2.0) and CHiME-6 (CC-BY-SA-4.0) are open.
If you redistribute audio extracted from this dataset, you must comply with
M3SD's and MSDWILD's non-commercial restriction.
## Citation
If you use this corpus, please cite the four source papers:
```bibtex
@article{wu2025m3sd,
title={M3SD: Multi-modal, Multi-scenario and Multi-language Speaker
Diarization Dataset},
author={Wu, Shilong and others},
journal={arXiv preprint arXiv:2506.14427},
year={2025}
}
@inproceedings{fu2021aishell4,
title={AISHELL-4: An Open Source Dataset for Speech Enhancement, Separation,
Recognition and Speaker Diarization in Conference Scenario},
author={Fu, Yihui and others},
booktitle={Interspeech},
year={2021}
}
@inproceedings{liu2022msdwild,
title={MSDWILD: Multi-modal Speaker Diarization Dataset in the Wild},
author={Liu, Tao and others},
booktitle={Interspeech},
year={2022}
}
@inproceedings{watanabe2020chime6,
title={CHiME-6 Challenge: Tackling Multispeaker Speech Recognition for
Unsegmented Recordings},
author={Watanabe, Shinji and others},
booktitle={CHiME Workshop},
year={2020}
}
```
ASR re-labelling was performed with [Qwen3-ASR-1.7B](https://huggingface.co/Qwen/Qwen3-ASR-1.7B).
## Loading
```python
from datasets import load_dataset
ds = load_dataset("ChristianYang/env-tts-sd-corpus", split="train", streaming=True)
row = next(iter(ds))
print(row["text"])
print(row["speech"]["sampling_rate"], len(row["speech"]["array"]))
```
The audio columns are typed as the HF `Audio` feature (16 kHz, mono), so
they decode automatically on access.
## Files on disk
```
data/
group_00000/
manifest.json
data_000000.parquet
data_000001.parquet
data_000002.parquet
data_000003.parquet
group_00001/
...
```
Each group is one atomic HF commit. Each parquet shard is ≈800 rows; group
size is 4 × 800 = 3 200 rows ≈ 250 MB (snappy-compressed, audio columns
already FLAC).
---
*Source pipeline: <https://github.com/...> (see the linked repository for
the streaming download/process/upload code that produced this dataset).*