Instructions to use Splintir/mms-tts-ceb-pld with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use Splintir/mms-tts-ceb-pld with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("text-to-audio", model="Splintir/mms-tts-ceb-pld")# Load model directly from transformers import AutoTokenizer, AutoModelForTextToWaveform tokenizer = AutoTokenizer.from_pretrained("Splintir/mms-tts-ceb-pld") model = AutoModelForTextToWaveform.from_pretrained("Splintir/mms-tts-ceb-pld", device_map="auto") - Notebooks
- Google Colab
- Kaggle
File size: 10,452 Bytes
20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 0e20360 20646b1 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 | """Turn one PLD speaker into a VITS training set, cached on the Hub.
python scripts/vits_data.py --language ceb --survey
python scripts/vits_data.py --language ceb --speaker-id top --push-to Splintir/pld-ceb-vits
Why this exists separately from `tts_data.py`: that script computes log-mel
spectrograms and x-vectors, because SpeechT5 consumes both. VITS consumes raw
waveform and text and nothing else -- it learns its own alignment and carries
one baked-in voice, so there is no speaker embedding to compute.
**Single speaker, deliberately.** MMS/VITS checkpoints hold exactly one voice.
Finetuning a one-voice model on PLD's many speakers averages them into mush,
which is the most likely reason the single-speaker SpeechT5 `-solo` run beat the
full-corpus `-v2` run on every statistic. Applying that lesson before the run
this time rather than after it.
`--survey` prints the speaker distribution and exits, so the choice of speaker
is made against clip counts and total duration rather than assumed. A VITS
finetune wants tens of minutes at minimum; if the top speaker is thin, the
survey says so before any GPU time is spent.
The scan reuses `tts_data.py`'s shard iterator: PLD's train split is 201 shards
of every language interleaved (~24 GB), so one language means touching all of
them, one shard on disk at a time.
"""
from __future__ import annotations
import argparse
import json
import os
import re
import sys
from collections import Counter, defaultdict
from pathlib import Path
sys.path.insert(0, str(Path(__file__).resolve().parent))
from tts_data import PLD_REPO, iter_shard_rows # noqa: E402
DIGIT_RE = re.compile(r"\d")
# MMS tokenizers are character-level and lowercase, with a per-language vocab
# that excludes digits. Rejecting a row is honest; keeping it would teach the
# model that "1990" is silence.
KEEP_RE = re.compile(r"[^a-zñáéíóú' -]")
def clean_text(text: str) -> str | None:
text = text.replace("’", "'").replace("‘", "'")
text = text.lower().strip()
if not text or DIGIT_RE.search(text):
return None
text = KEEP_RE.sub("", text)
text = re.sub(r"\s+", " ", text).strip()
return text or None
def survey(language: str, token: str | None, shards: int) -> None:
"""Print who speaks this language and for how long, then stop."""
clips: Counter[str] = Counter()
secs: defaultdict[str, float] = defaultdict(float)
# gender/age are not in tts_data.COLUMNS and the iterator only reads those,
# so the survey reports clips and duration -- the two numbers that decide
# whether a speaker can carry a finetune.
for n, (_, row) in enumerate(iter_shard_rows(language, token, range(shards)), 1):
sid = row["speaker_id"]
clips[sid] += 1
secs[sid] += float(row.get("duration") or 0.0)
if n % 2000 == 0:
print(f" ... {n} rows, {len(clips)} speakers", flush=True)
total = sum(clips.values())
print(f"\n{language}: {total} usable clips, {len(clips)} speakers, "
f"{sum(secs.values()) / 3600:.1f} h total\n")
print(f"{'speaker_id':<26}{'clips':>8}{'minutes':>10}")
for sid, n in clips.most_common(20):
print(f"{sid:<26}{n:>8}{secs[sid] / 60:>10.1f}")
print("\nA VITS finetune wants >= ~30 min from one speaker. Pick from the "
"top rows and rerun with --speaker-id.")
def cache_dir(language: str, speaker: str) -> Path:
return Path("vits_cache") / language / speaker
def load_cached(language: str, speaker: str):
"""Reuse a completed scan. Scanning 201 shards to find ~15 minutes of audio
costs ~24 GB of transfer, so it must never be repeated because a later step
failed."""
manifest = cache_dir(language, speaker) / "manifest.jsonl"
if not manifest.exists():
return None
rows = [json.loads(line) for line in
manifest.read_text(encoding="utf-8").splitlines() if line.strip()]
rows = [r for r in rows if (cache_dir(language, speaker) / r["file"]).exists()]
if not rows:
return None
print(f"reusing {len(rows)} cached clips from "
f"{cache_dir(language, speaker)} (delete it to force a rescan)",
flush=True)
return rows
def collect(language: str, speaker: str, token: str | None, shards: int,
max_seconds: float):
"""Gather one speaker's clips. Returns (records, resolved_speaker_id)."""
import io
import soundfile as sf
# `top` cannot be resolved until the corpus has been scanned once, so the
# first pass counts and the second keeps. Two passes over 24 GB is slow;
# buffering every language's audio in RAM instead is worse.
if speaker == "top":
counts: Counter[str] = Counter()
for _, row in iter_shard_rows(language, token, range(shards)):
counts[row["speaker_id"]] += 1
if not counts:
raise SystemExit(f"no usable {language} rows -- check the filters")
speaker, n = counts.most_common(1)[0]
print(f"resolved `top` -> {speaker} ({n} clips)", flush=True)
out = cache_dir(language, speaker)
out.mkdir(parents=True, exist_ok=True)
manifest = (out / "manifest.jsonl").open("w", encoding="utf-8")
records, total = [], 0.0
seen_shard = -1
for shard, row in iter_shard_rows(language, token, range(shards)):
# One speaker is a handful of clips scattered over 201 shards, so a
# clip-count progress line can stay silent for hours. Report the scan
# itself instead -- otherwise a live run is indistinguishable from a
# hung one.
if shard != seen_shard:
seen_shard = shard
print(f" shard {shard + 1}/{shards} kept {len(records)} clips, "
f"{total / 60:.1f} min", flush=True)
if row["speaker_id"] != speaker:
continue
text = clean_text(row["sentence"])
if not text:
continue
raw = row["audio"]["bytes"]
# Decode once here rather than trusting the shard's declared duration:
# the trainer segments on real sample counts, and a mismatch shows up as
# a silent crash deep in the collator.
try:
wav, rate = sf.read(io.BytesIO(raw), dtype="float32", always_2d=False)
except Exception as exc: # noqa: BLE001
print(f" skipped unreadable clip: {exc}", flush=True)
continue
if wav.ndim > 1:
wav = wav.mean(axis=1)
secs = len(wav) / rate
if not 1.0 <= secs <= 15.0:
continue
# Write PLD's own encoded bytes straight through rather than re-encoding
# the decoded array: no quality loss, and `datasets` can build an Audio
# column from encoded bytes without torchcodec, which it needs for raw
# arrays and file paths alike.
ext = Path(row["audio"].get("path") or "clip.wav").suffix or ".wav"
name = f"{len(records):04d}{ext}"
(out / name).write_bytes(raw)
manifest.write(json.dumps({"file": name, "text": text,
"seconds": round(secs, 3)}) + "\n")
manifest.flush()
records.append({"file": name, "text": text, "seconds": round(secs, 3)})
total += secs
if max_seconds and total >= max_seconds:
break
manifest.close()
print(f"\n{speaker}: {len(records)} clips, {total / 60:.1f} min "
f"-> cached in {out}", flush=True)
if total < 900:
print("WARNING: under 15 minutes. Expect a weak finetune -- consider "
"pooling a second speaker of the same gender and dialect.",
flush=True)
return records, speaker
def main() -> None:
ap = argparse.ArgumentParser(description=__doc__.split("\n")[0])
ap.add_argument("--language", default="ceb", help="PLD ISO 639-3 code")
ap.add_argument("--speaker-id", default="top",
help="`top` picks the speaker with the most clips")
ap.add_argument("--survey", action="store_true",
help="print the speaker distribution and exit")
ap.add_argument("--shards", type=int, default=201)
ap.add_argument("--max-seconds", type=float, default=0,
help="stop after this much audio (0 = no cap)")
ap.add_argument("--push-to", default="",
help="Hub dataset repo, e.g. Splintir/pld-ceb-vits")
args = ap.parse_args()
token = os.environ.get("HF_TOKEN")
if not token:
env = Path(__file__).resolve().parent.parent / ".env"
if env.exists():
for line in env.read_text(encoding="utf-8").splitlines():
key, _, value = line.strip().partition("=")
if key == "HF_TOKEN" and value:
token = value.strip()
print(f"scanning {args.shards} {PLD_REPO} train shards for {args.language} ...",
flush=True)
if args.survey:
survey(args.language, token, args.shards)
return
speaker = args.speaker_id
records = None if speaker == "top" else load_cached(args.language, speaker)
if records is None:
records, speaker = collect(args.language, args.speaker_id, token,
args.shards, args.max_seconds)
if not records:
raise SystemExit(f"no clips for speaker {speaker}")
from datasets import Audio, Dataset
src = cache_dir(args.language, speaker)
rows = [{"audio": {"bytes": (src / r["file"]).read_bytes(),
"path": r["file"]},
"text": r["text"]}
for r in records]
ds = Dataset.from_list(rows).cast_column("audio", Audio(sampling_rate=16000))
# A held-out slice the trainer can score against, kept small: VITS eval is
# slow (it renders audio) and the number that decides anything is the
# 50-line bench, not this. Proportional with a floor and a ceiling -- a
# flat floor alone puts more clips in eval than train on a small set.
n_eval = max(4, min(16, len(ds) // 10))
ds = ds.train_test_split(test_size=n_eval, seed=0)
print(ds)
if args.push_to:
ds.push_to_hub(args.push_to, token=token, private=True)
print(f"pushed -> {args.push_to} (speaker {speaker})")
else:
out = Path("vits_data") / args.language
ds.save_to_disk(str(out))
print(f"saved -> {out} (pass --push-to to upload)")
if __name__ == "__main__":
main()
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