File size: 5,788 Bytes
d8274ac | 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 | """Can whispering be instructed instead of cloned?
Six low-voicing clips all failed as prompts: either the clone lost the breath
(buy1take2, voicing 0.34 -> 0.71) or it kept the breath and invented words
(Rainnight, CER 0.53). Whisper has no F0, and the model's voice conditioning
leans on periodicity, so a 6 s breathy prompt is not enough to carry it.
The repo exposes two text-side controls that were never tried here:
FREEFORM a free-text instruction, e.g. "用气声耳语,很轻"
BASIC speed/pitch/volume from fixed vocabularies
Both keep a *working* prompt (a08) and move the style into the text, which is
the opposite of what we were doing. Success means low CER *and* low voicing of
the output - a soft normal voice is not a whisper.
Writes style_probe/*.wav and metrics.json.
"""
import json
import os
import sys
import time
import traceback
from pathlib import Path
import numpy as np
ROOT = Path(os.environ.get("WORK_ROOT", "/kaggle/working"))
REPO = ROOT / "Ming-3.0-tts"
MODEL = ROOT / "Ming-3.0-tts-0.5b"
BUNDLE = ROOT / "webfm-eval"
OUT = ROOT / "style_probe"
SEED = 20261031
VOICE = "a08"
LINES = ["后来店关了,歌还在。", "晚安,我们明天见。", "别急,先呼吸。"]
TRIALS = [
("plain", {}),
("freeform_whisper", {"task": "FREEFORM",
"instruction": "用气声耳语,很轻,像在深夜贴着耳朵说话。"}),
("freeform_soft", {"task": "FREEFORM",
"instruction": "轻声说,音量很低,气息多,接近耳语。"}),
("basic_low", {"task": "BASIC", "speed": "慢速", "pitch": "低", "volume": "低"}),
("freeform_basic", {"task": "FREEFORM",
"instruction": "用耳语说,很慢,很低,几乎只剩气息。"}),
]
def log(msg: str) -> None:
print(f"[{time.strftime('%H:%M:%S')}] {msg}", flush=True)
def patch_fp32() -> None:
for name in ("ming_tts.py", "modeling_bailingmm.py"):
p = REPO / name
s = p.read_text()
s = s.replace("torch_dtype=torch.bfloat16", "torch_dtype=torch.float32")
s = s.replace(".eval().to(torch.bfloat16).to(device)", ".eval().to(torch.float32).to(device)")
s = s.replace("dtype=torch.bfloat16", "dtype=torch.float32")
p.write_text(s)
def voicing(y, sr):
win, hop = int(0.04 * sr), int(0.02 * sr)
if len(y) < win * 2:
return 0.0
frames = np.lib.stride_tricks.sliding_window_view(y, win)[::hop]
energy = (frames ** 2).mean(1)
idx = np.argsort(energy)[-max(1, int(len(energy) * 0.3)):]
lo, hi = int(sr / 400), int(sr / 70)
peaks = []
for f in frames[idx]:
f = f - f.mean()
ac = np.correlate(f, f, "full")[win - 1:]
if ac[0] > 0:
peaks.append(float(ac[lo:hi].max() / ac[0]))
return float(np.median(peaks)) if peaks else 0.0
def main() -> None:
import torch
import soundfile as sf
from transformers import pipeline
OUT.mkdir(parents=True, exist_ok=True)
sys.path.insert(0, str(REPO))
patch_fp32()
from ming_tts import MingTTS
import tts_tasks
sys.path.insert(0, str(BUNDLE / "eval"))
from score import load_t2s, score_row
table = load_t2s()
voices = {v["slot"]: v for v in
json.loads((BUNDLE / "mined/manifest.json").read_text())["voices"]}
v = voices[VOICE]
tts = MingTTS(str(MODEL), device="cuda", text_normalization=False)
rows = []
for name, kw in TRIALS:
item = {"trial": name, "kwargs": {k: str(x) for k, x in kw.items()}}
ws, vs = [], []
for i, text in enumerate(LINES, 1):
try:
torch.manual_seed(SEED + i)
w = tts.synthesize(text=text, language_id="Chinese",
prompt_wav_path=str(BUNDLE / v["wav"]),
prompt_text=v["prompt_text"], use_spk_emb=True,
raw_text=True,
max_decode_steps=max(32, min(400, int(len(text) * 3.5))),
**kw)
y = w.cpu().numpy().reshape(-1)
sf.write(str(OUT / f"{name}_{i}.wav"), y, 44100)
ws.append(round(y.size / 44100, 2))
vs.append(voicing(y, 44100))
except Exception:
log(f"{name} line {i} failed\n{traceback.format_exc()}")
item["seconds"] = ws
item["voicing"] = float(np.mean(vs)) if vs else 0.0
rows.append(item)
log(f"{name}: v_out={item['voicing']:.2f} sec={ws}")
asr = pipeline("automatic-speech-recognition", model="openai/whisper-small",
device=0, torch_dtype=torch.float16)
for r in rows:
ces = []
for i, text in enumerate(LINES, 1):
p = OUT / f"{r['trial']}_{i}.wav"
if not p.exists():
continue
y, _ = sf.read(p, dtype="float32")
out = asr({"array": y.astype(np.float32), "sampling_rate": 44100},
generate_kwargs={"language": "zh", "task": "transcribe"})
hyp = (out.get("text", "") if isinstance(out, dict) else str(out)).strip()
ces.append(score_row({"text": text, "hyp_text": hyp}, table)["cer_norm"])
r[f"hyp_{i}"] = hyp
r["cer"] = float(np.mean(ces)) if ces else 1.0
log(f"{r['trial']} cer={r['cer']:.3f} v={r['voicing']:.2f} {r.get('hyp_1','')[:22]}")
rows.sort(key=lambda r: r["cer"])
(OUT / "metrics.json").write_text(json.dumps(rows, ensure_ascii=False, indent=1))
log("\n-- summary (CER low is good; voicing low means breathy) --")
for r in rows:
log(f" cer={r['cer']:.2f} v={r['voicing']:.2f} {r['trial']}")
if __name__ == "__main__":
main()
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