lyric-sync / app.py
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feat: use `htdemucs` model for stem separation instead of `htdemucs_ft`
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"""
🎡 Lyric Sync β€” Gradio Space
Automatic perfect song lyric acquisition and synchronization.
Pipeline:
1. Vocal separation (Demucs htdemucs)
2. Word-level transcription (Whisper large-v3)
3. Song identification (AcoustID / transcript search)
4. Lyrics fetching (LRCLIB)
5. Sequence alignment (transfer timings to correct lyrics)
6. Timing refinement (onset/offset detection)
"""
import json
import logging
import os
import re
import subprocess
import tempfile
import time
import unicodedata
from dataclasses import dataclass, field
from difflib import SequenceMatcher
from pathlib import Path
from typing import Optional
import gradio as gr
import librosa
import numpy as np
import requests
import spaces
import torch
import torchaudio
logging.basicConfig(level=logging.INFO, format="%(asctime)s [%(levelname)s] %(message)s")
logger = logging.getLogger(__name__)
# ═══════════════════════════════════════════════════════════════════════════════
# AUDIO LOADING (ffmpeg-based, handles all formats)
# ═══════════════════════════════════════════════════════════════════════════════
def load_audio_as_tensor(audio_path: str, target_sr: int = 44100) -> tuple[torch.Tensor, int]:
"""
Load any audio file into a torch tensor using librosa (ffmpeg backend).
Returns: (tensor [channels, samples], sample_rate)
Always returns stereo at target_sr.
"""
# librosa.load handles MP3, FLAC, OGG, WAV, etc. via soundfile/ffmpeg
y, sr = librosa.load(audio_path, sr=target_sr, mono=False)
# y shape: (samples,) if mono or (channels, samples) if stereo
if y.ndim == 1:
# Mono β†’ stereo by duplicating
wav = torch.from_numpy(y).float().unsqueeze(0).repeat(2, 1)
else:
wav = torch.from_numpy(y).float()
if wav.shape[0] > 2:
wav = wav[:2]
return wav, target_sr
# ═══════════════════════════════════════════════════════════════════════════════
# DATA CLASSES
# ═══════════════════════════════════════════════════════════════════════════════
@dataclass
class TimedWord:
word: str
start: float
end: float
confidence: float = 1.0
@property
def duration(self) -> float:
return self.end - self.start
@dataclass
class SongInfo:
title: str
artist: str
album: Optional[str] = None
method: str = "unknown"
# ═══════════════════════════════════════════════════════════════════════════════
# STEP 1: VOCAL SEPARATION (Demucs)
# ═══════════════════════════════════════════════════════════════════════════════
_demucs_model = None
def get_demucs_model(device="cpu"):
global _demucs_model
if _demucs_model is None:
from demucs.pretrained import get_model
_demucs_model = get_model("htdemucs")
_demucs_model.eval()
logger.info("Demucs htdemucs loaded")
_demucs_model.to(device)
return _demucs_model
def separate_vocals(audio_path: str, device: str = "cpu") -> tuple[np.ndarray, int, np.ndarray, int]:
"""
Separate vocals from audio.
Returns: (vocals_16k_mono, 16000, vocals_44k_mono, 44100)
"""
from demucs.apply import apply_model
model = get_demucs_model(device)
# Load audio using librosa (handles MP3, FLAC, OGG, etc.)
wav, sr = load_audio_as_tensor(audio_path, target_sr=44100)
# Run separation
wav_batch = wav.unsqueeze(0).to(device)
with torch.no_grad():
sources = apply_model(
model, wav_batch, device=device,
shifts=1, split=True, overlap=0.25, progress=False,
)
# sources: [1, 4, 2, N] β€” drums, bass, other, vocals
vocal_idx = model.sources.index("vocals")
vocals_stereo = sources[0, vocal_idx].cpu() # [2, N] at 44100
vocals_44k = vocals_stereo.mean(dim=0) # mono
# Resample to 16kHz for Whisper
vocals_16k = torchaudio.functional.resample(
vocals_44k.unsqueeze(0), 44100, 16000
).squeeze(0)
return vocals_16k.numpy(), 16000, vocals_44k.numpy(), 44100
# ═══════════════════════════════════════════════════════════════════════════════
# STEP 2: TRANSCRIPTION (Whisper)
# ═══════════════════════════════════════════════════════════════════════════════
_whisper_pipe = None
def get_whisper_pipeline(device="cpu"):
global _whisper_pipe
if _whisper_pipe is None:
from transformers import pipeline as hf_pipeline
_whisper_pipe = hf_pipeline(
task="automatic-speech-recognition",
model="openai/whisper-large-v3",
dtype=torch.float16 if device == "cuda" else torch.float32,
device=device,
model_kwargs={"attn_implementation": "sdpa"},
)
logger.info(f"Whisper large-v3 pipeline loaded on {device}")
return _whisper_pipe
def transcribe_vocals(audio: np.ndarray, sr: int = 16000) -> list[TimedWord]:
pipe = get_whisper_pipeline("cuda" if torch.cuda.is_available() else "cpu")
result = pipe(
{"array": audio.astype(np.float32), "sampling_rate": sr},
return_timestamps="word",
generate_kwargs={
"language": "english",
"task": "transcribe",
},
chunk_length_s=30,
stride_length_s=5,
)
words = []
for chunk in result.get("chunks", []):
text = chunk["text"].strip()
ts = chunk.get("timestamp", (None, None))
if text and ts[0] is not None and ts[1] is not None:
words.append(TimedWord(word=text, start=ts[0], end=ts[1]))
return words
# ═══════════════════════════════════════════════════════════════════════════════
# STEP 3: SONG IDENTIFICATION
# ═══════════════════════════════════════════════════════════════════════════════
def identify_song(audio_path, user_artist="", user_title="", transcript_text=""):
if user_artist.strip() and user_title.strip():
return SongInfo(title=user_title.strip(), artist=user_artist.strip(), method="user_provided")
acoustid_key = os.environ.get("ACOUSTID_API_KEY")
if acoustid_key:
result = _acoustid_identify(audio_path, acoustid_key)
if result:
return result
if transcript_text:
words = transcript_text.split()
if len(words) >= 5:
mid = len(words) // 2
fragment = " ".join(words[max(0, mid-5):mid+5])
result = _search_lrclib(fragment)
if result:
return result
return None
def _acoustid_identify(audio_path, api_key):
try:
result = subprocess.run(
["fpcalc", "-json", "-length", "120", audio_path],
capture_output=True, text=True, timeout=30,
)
if result.returncode != 0:
return None
fp_data = json.loads(result.stdout)
resp = requests.post("https://api.acoustid.org/v2/lookup", data={
"client": api_key, "duration": fp_data["duration"],
"fingerprint": fp_data["fingerprint"],
"meta": "recordings releasegroups", "format": "json",
}, timeout=15)
resp.raise_for_status()
data = resp.json()
if data.get("status") != "ok" or not data.get("results"):
return None
best = max(data["results"], key=lambda r: r.get("score", 0))
if best.get("score", 0) < 0.5 or not best.get("recordings"):
return None
rec = best["recordings"][0]
artist = rec.get("artists", [{}])[0].get("name", "Unknown")
album = rec.get("releasegroups", [{}])[0].get("title") if rec.get("releasegroups") else None
return SongInfo(title=rec.get("title", "Unknown"), artist=artist, album=album, method="acoustid")
except Exception as e:
logger.warning(f"AcoustID failed: {e}")
return None
def _search_lrclib(query):
try:
resp = requests.get("https://lrclib.net/api/search", params={"q": query}, timeout=10)
if resp.status_code == 200:
results = resp.json()
if results:
best = results[0]
return SongInfo(title=best.get("trackName", "Unknown"), artist=best.get("artistName", "Unknown"), album=best.get("albumName"), method="lrclib_search")
except Exception as e:
logger.debug(f"LRCLIB search failed: {e}")
return None
# ═══════════════════════════════════════════════════════════════════════════════
# STEP 4: LYRICS FETCHING
# ═══════════════════════════════════════════════════════════════════════════════
def fetch_lyrics(song):
params = {"artist_name": song.artist, "track_name": song.title}
if song.album:
params["album_name"] = song.album
try:
resp = requests.get("https://lrclib.net/api/get", params=params, timeout=10)
if resp.status_code == 200:
data = resp.json()
plain = data.get("plainLyrics") or data.get("syncedLyrics", "")
if plain:
plain = re.sub(r"\[\d{2}:\d{2}\.\d{2,3}\]\s*", "", plain)
words = plain.split()
if words:
return plain, words
except Exception as e:
logger.warning(f"LRCLIB fetch failed: {e}")
try:
resp = requests.get("https://lrclib.net/api/search", params={"q": f"{song.artist} {song.title}"}, timeout=10)
if resp.status_code == 200:
results = resp.json()
if results:
plain = results[0].get("plainLyrics") or results[0].get("syncedLyrics", "")
plain = re.sub(r"\[\d{2}:\d{2}\.\d{2,3}\]\s*", "", plain)
words = plain.split()
if words:
return plain, words
except Exception as e:
logger.debug(f"LRCLIB search fallback failed: {e}")
return None
# ═══════════════════════════════════════════════════════════════════════════════
# STEP 5: SEQUENCE ALIGNMENT
# ═══════════════════════════════════════════════════════════════════════════════
def normalize_word(word):
word = unicodedata.normalize("NFKD", word).lower()
word = re.sub(r"[^\w']", "", word).strip("'")
return word
def align_words(asr_words, ref_words):
if not asr_words or not ref_words:
return asr_words or []
asr_norm = [normalize_word(w.word) for w in asr_words]
ref_norm = [normalize_word(w) for w in ref_words]
asr_valid = [(i, n) for i, n in enumerate(asr_norm) if n]
ref_valid = [(i, n) for i, n in enumerate(ref_norm) if n]
if not asr_valid or not ref_valid:
return asr_words
asr_indices, asr_normalized = zip(*asr_valid)
ref_indices, ref_normalized = zip(*ref_valid)
ref_set = set(ref_normalized)
asr_for_matching = []
for aw in asr_normalized:
if aw in ref_set:
asr_for_matching.append(aw)
else:
best_match, best_ratio = aw, 0.0
for rw in ref_set:
if abs(len(aw) - len(rw)) > max(len(aw), len(rw)) * 0.4:
continue
ratio = SequenceMatcher(None, aw, rw).ratio()
if ratio > best_ratio:
best_ratio, best_match = ratio, rw
asr_for_matching.append(best_match if best_ratio >= 0.75 else aw)
sm = SequenceMatcher(None, asr_for_matching, list(ref_normalized), autojunk=False)
opcodes = sm.get_opcodes()
result = [TimedWord(word=w, start=0.0, end=0.0, confidence=0.0) for w in ref_words]
for tag, i1, i2, j1, j2 in opcodes:
if tag == "equal":
for asr_pos, ref_pos in zip(range(i1, i2), range(j1, j2)):
orig_asr, orig_ref = asr_indices[asr_pos], ref_indices[ref_pos]
result[orig_ref] = TimedWord(word=ref_words[orig_ref], start=asr_words[orig_asr].start, end=asr_words[orig_asr].end, confidence=asr_words[orig_asr].confidence)
elif tag == "replace":
t_start = asr_words[asr_indices[i1]].start
t_end = asr_words[asr_indices[i2 - 1]].end
n_ref = j2 - j1
duration = t_end - t_start
for k, ref_pos in enumerate(range(j1, j2)):
orig_ref = ref_indices[ref_pos]
result[orig_ref] = TimedWord(word=ref_words[orig_ref], start=t_start + k * duration / n_ref, end=t_start + (k + 1) * duration / n_ref, confidence=0.5)
_fill_gaps(result)
return result
def _fill_gaps(words):
for i in range(len(words)):
if words[i].start > 0 or words[i].end > 0:
continue
prev_end, next_start = 0.0, None
for j in range(i - 1, -1, -1):
if words[j].end > 0:
prev_end = words[j].end
break
for j in range(i + 1, len(words)):
if words[j].start > 0:
next_start = words[j].start
break
if next_start is None:
next_start = prev_end + 0.3
gap_count = 0
for j in range(i, len(words)):
if words[j].start > 0 or words[j].end > 0:
break
gap_count += 1
position = 0
for j in range(i - 1, -1, -1):
if words[j].start > 0 or words[j].end > 0:
break
position += 1
t_per_word = (next_start - prev_end) / max(gap_count, 1)
words[i].start = prev_end + position * t_per_word
words[i].end = prev_end + (position + 1) * t_per_word
words[i].confidence = 0.3
# ═══════════════════════════════════════════════════════════════════════════════
# STEP 6: TIMING REFINEMENT
# ═══════════════════════════════════════════════════════════════════════════════
def refine_timings(vocals_44k, words, sr=44100, hop_length=256):
if len(vocals_44k) == 0 or not words:
return words
odf = librosa.onset.onset_strength(y=vocals_44k, sr=sr, hop_length=hop_length, n_fft=1024, fmin=80.0, fmax=4000.0, aggregate=np.median, detrend=True)
onset_frames = librosa.onset.onset_detect(onset_envelope=odf, sr=sr, hop_length=hop_length, backtrack=True, units='frames', pre_max=2, post_max=2, pre_avg=2, post_avg=4, delta=0.05, wait=8)
rms = librosa.feature.rms(y=vocals_44k, frame_length=1024, hop_length=hop_length)[0]
rms_smooth = np.convolve(rms, np.ones(7) / 7, mode='same')
search_frames = int(0.08 * sr / hop_length)
refined = []
for word in words:
w = TimedWord(word=word.word, start=word.start, end=word.end, confidence=word.confidence)
approx_frame = librosa.time_to_frames(w.start, sr=sr, hop_length=hop_length)
lo, hi = max(0, approx_frame - search_frames), min(len(odf) - 1, approx_frame + search_frames)
candidates = onset_frames[(onset_frames >= lo) & (onset_frames <= hi)]
if len(candidates) > 0:
w.start = librosa.frames_to_time(candidates[np.argmin(np.abs(candidates - approx_frame))], sr=sr, hop_length=hop_length)
end_frame = librosa.time_to_frames(w.end, sr=sr, hop_length=hop_length)
end_search = int(0.05 * sr / hop_length)
elo, ehi = max(0, end_frame - end_search), min(len(rms_smooth) - 1, end_frame + end_search)
if elo < ehi:
rms_db = librosa.amplitude_to_db(rms_smooth[elo:ehi + 1] + 1e-10, ref=rms_smooth.max() + 1e-10)
silent = np.where(rms_db < -40.0)[0]
if len(silent) > 0:
w.end = librosa.frames_to_time(elo + silent[0], sr=sr, hop_length=hop_length)
if w.end <= w.start + 0.03:
w.end = w.start + 0.03
refined.append(w)
for i in range(len(refined) - 1):
if refined[i].end > refined[i + 1].start:
mid = (refined[i].end + refined[i + 1].start) / 2
refined[i] = TimedWord(word=refined[i].word, start=refined[i].start, end=mid, confidence=refined[i].confidence)
refined[i + 1] = TimedWord(word=refined[i + 1].word, start=mid, end=refined[i + 1].end, confidence=refined[i + 1].confidence)
return refined
# ═══════════════════════════════════════════════════════════════════════════════
# OUTPUT FORMATTERS
# ═══════════════════════════════════════════════════════════════════════════════
def _lrc_ts(seconds):
return f"{int(seconds // 60):02d}:{seconds % 60:05.2f}"
def _srt_ts(seconds):
h, m, s = int(seconds // 3600), int((seconds % 3600) // 60), seconds % 60
return f"{h:02d}:{m:02d}:{int(s):02d},{int((s % 1) * 1000):03d}"
def format_enhanced_lrc(words, line_gap=1.0):
if not words: return ""
lines, current_line = [], []
for word in words:
if current_line and word.start - current_line[-1].end > line_gap:
lines.append(f"[{_lrc_ts(current_line[0].start)}] " + " ".join(f"<{_lrc_ts(w.start)}> {w.word}" for w in current_line) + f" <{_lrc_ts(current_line[-1].end)}>")
current_line = []
current_line.append(word)
if current_line:
lines.append(f"[{_lrc_ts(current_line[0].start)}] " + " ".join(f"<{_lrc_ts(w.start)}> {w.word}" for w in current_line) + f" <{_lrc_ts(current_line[-1].end)}>")
return "\n".join(lines)
def format_standard_lrc(words, line_gap=1.0):
if not words: return ""
lines, current_line = [], []
for word in words:
if current_line and word.start - current_line[-1].end > line_gap:
lines.append(f"[{_lrc_ts(current_line[0].start)}] " + " ".join(w.word for w in current_line))
current_line = []
current_line.append(word)
if current_line:
lines.append(f"[{_lrc_ts(current_line[0].start)}] " + " ".join(w.word for w in current_line))
return "\n".join(lines)
def format_srt(words, line_gap=1.0, max_words=10):
if not words: return ""
entries, current_line = [], []
for word in words:
if current_line and (word.start - current_line[-1].end > line_gap or len(current_line) >= max_words):
entries.append(current_line)
current_line = []
current_line.append(word)
if current_line: entries.append(current_line)
return "\n".join(f"{idx}\n{_srt_ts(lw[0].start)} --> {_srt_ts(lw[-1].end)}\n{' '.join(w.word for w in lw)}\n" for idx, lw in enumerate(entries, 1))
def format_json(words):
return [{"word": w.word, "start": round(w.start, 3), "end": round(w.end, 3), "confidence": round(w.confidence, 3)} for w in words]
def format_ass(words, line_gap=1.0):
header = "[Script Info]\nTitle: Lyric Sync\nScriptType: v4.00+\nPlayResX: 1920\nPlayResY: 1080\n\n[V4+ Styles]\nFormat: Name, Fontname, Fontsize, PrimaryColour, SecondaryColour, OutlineColour, BackColour, Bold, Italic, Underline, StrikeOut, ScaleX, ScaleY, Spacing, Angle, BorderStyle, Outline, Shadow, Alignment, MarginL, MarginR, MarginV, Encoding\nStyle: Default,Arial,48,&H00FFFFFF,&H000000FF,&H00000000,&H64000000,-1,0,0,0,100,100,0,0,1,2,1,2,10,10,40,1\n\n[Events]\nFormat: Layer, Start, End, Style, Name, MarginL, MarginR, MarginV, Effect, Text\n"
line_groups, current = [], []
for word in words:
if current and word.start - current[-1].end > line_gap:
line_groups.append(current)
current = []
current.append(word)
if current: line_groups.append(current)
def ass_ts(s): return f"{int(s//3600)}:{int((s%3600)//60):02d}:{s%60:05.2f}"
events = []
for lw in line_groups:
karaoke = " ".join(f"{{\\kf{int(w.duration * 100)}}}{w.word}" for w in lw)
events.append(f"Dialogue: 0,{ass_ts(lw[0].start)},{ass_ts(lw[-1].end)},Default,,0,0,0,,{karaoke}")
return header + "\n".join(events)
# ═══════════════════════════════════════════════════════════════════════════════
# MAIN PIPELINE
# ═══════════════════════════════════════════════════════════════════════════════
@spaces.GPU(duration=300)
def run_pipeline(audio_path: str, artist: str, title: str, progress=gr.Progress()):
if audio_path is None:
raise gr.Error("Please upload an audio file.")
start_time = time.time()
status_log = []
def log(msg):
status_log.append(msg)
logger.info(msg)
device = "cuda" if torch.cuda.is_available() else "cpu"
log(f"πŸ–₯️ Using device: {device}")
progress(0.05, desc="🎡 Separating vocals (Demucs htdemucs)...")
log("🎡 Separating vocals with Demucs htdemucs...")
vocals_16k, sr_16k, vocals_44k, sr_44k = separate_vocals(audio_path, device=device)
log(f" βœ“ Vocals: {len(vocals_16k)/sr_16k:.1f}s extracted")
progress(0.30, desc="πŸ“ Transcribing vocals (Whisper large-v3)...")
log("πŸ“ Transcribing with Whisper large-v3 (word-level timestamps)...")
asr_words = transcribe_vocals(vocals_16k, sr_16k)
transcript_text = " ".join(w.word for w in asr_words)
log(f" βœ“ Transcribed: {len(asr_words)} words")
progress(0.55, desc="πŸ” Identifying song...")
log("πŸ” Identifying song...")
song = identify_song(audio_path, artist, title, transcript_text)
if song:
log(f" βœ“ Identified: {song.artist} β€” {song.title} (via {song.method})")
else:
log(" ⚠ Could not identify song β€” using raw transcription")
ref_words = None
if song:
progress(0.65, desc="πŸ“– Fetching reference lyrics (LRCLIB)...")
log("πŸ“– Fetching lyrics from LRCLIB...")
lyrics_result = fetch_lyrics(song)
if lyrics_result:
_, ref_words = lyrics_result
log(f" βœ“ Lyrics: {len(ref_words)} words")
else:
log(" ⚠ No lyrics found β€” using raw transcription")
if ref_words:
progress(0.75, desc="πŸ”— Aligning transcript to reference lyrics...")
log("πŸ”— Aligning ASR transcript to reference lyrics...")
synced_words = align_words(asr_words, ref_words)
matched = sum(1 for w in synced_words if w.confidence >= 0.8)
total = len(synced_words)
log(f" βœ“ Aligned: {matched}/{total} direct matches ({matched/total*100:.0f}%)" if total else " βœ“ Aligned")
else:
synced_words = asr_words
log(" Using raw ASR words (no reference lyrics)")
progress(0.88, desc="βš™οΈ Refining timings (onset/offset detection)...")
log("βš™οΈ Refining timings via audio analysis (onset detection, librosa)...")
synced_words = refine_timings(vocals_44k, synced_words, sr=sr_44k)
log(" βœ“ Timing refinement complete")
progress(0.96, desc="πŸ“„ Formatting outputs...")
lrc_enhanced = format_enhanced_lrc(synced_words)
lrc_standard = format_standard_lrc(synced_words)
srt_text = format_srt(synced_words)
json_data = format_json(synced_words)
ass_text = format_ass(synced_words)
def write_tmp(content, suffix):
f = tempfile.NamedTemporaryFile(suffix=suffix, delete=False, mode="w", encoding="utf-8")
f.write(content); f.close(); return f.name
lrc_file = write_tmp(lrc_enhanced, ".lrc")
srt_file = write_tmp(srt_text, ".srt")
json_file = write_tmp(json.dumps(json_data, indent=2, ensure_ascii=False), ".json")
ass_file = write_tmp(ass_text, ".ass")
elapsed = time.time() - start_time
log(f"\nβœ… Done in {elapsed:.1f}s β€” {len(synced_words)} words synchronized")
progress(1.0, desc="βœ… Done!")
return ("\n".join(status_log), lrc_enhanced, lrc_file, lrc_standard, srt_text, srt_file, json_data, json_file, ass_text, ass_file)
# ═══════════════════════════════════════════════════════════════════════════════
# GRADIO UI
# ═══════════════════════════════════════════════════════════════════════════════
with gr.Blocks(title="🎡 Lyric Sync", css=".main-title{text-align:center;margin-bottom:.5em}.subtitle{text-align:center;color:#666;margin-bottom:1.5em}") as demo:
gr.HTML("<h1 class='main-title'>🎡 Lyric Sync</h1>")
gr.HTML("<p class='subtitle'>Automatic perfect song lyric acquisition and synchronization</p>")
with gr.Row():
with gr.Column(scale=1):
audio_input = gr.Audio(sources=["upload"], type="filepath", label="Upload Audio")
gr.Markdown("### Song Metadata (optional)\n*Provide artist & title to skip identification, or leave blank for auto-detection.*")
artist_input = gr.Textbox(label="Artist", placeholder="e.g. Radiohead")
title_input = gr.Textbox(label="Song Title", placeholder="e.g. Creep")
run_btn = gr.Button("β–Ά Sync Lyrics", variant="primary", size="lg")
status_output = gr.Textbox(label="Pipeline Status", lines=12, interactive=False, show_copy_button=True)
with gr.Column(scale=2):
with gr.Tabs():
with gr.Tab("πŸ“ Enhanced LRC (word-level)"):
lrc_enhanced_output = gr.Textbox(label="Enhanced LRC", lines=20, show_copy_button=True, interactive=False)
lrc_download = gr.File(label="⬇ Download .lrc")
with gr.Tab("πŸ“„ Standard LRC (line-level)"):
lrc_standard_output = gr.Textbox(label="Standard LRC", lines=20, show_copy_button=True, interactive=False)
with gr.Tab("🎬 SRT Subtitles"):
srt_output = gr.Textbox(label="SRT", lines=20, show_copy_button=True, interactive=False)
srt_download = gr.File(label="⬇ Download .srt")
with gr.Tab("πŸ“Š JSON"):
json_output = gr.JSON(label="Word-level JSON")
json_download = gr.File(label="⬇ Download .json")
with gr.Tab("🎀 ASS Karaoke"):
ass_output = gr.Textbox(label="ASS (karaoke \\k tags)", lines=20, show_copy_button=True, interactive=False)
ass_download = gr.File(label="⬇ Download .ass")
run_btn.click(fn=run_pipeline, inputs=[audio_input, artist_input, title_input], outputs=[status_output, lrc_enhanced_output, lrc_download, lrc_standard_output, srt_output, srt_download, json_output, json_download, ass_output, ass_download])
gr.Markdown("---\n### How it works\n| Step | What | Technology |\n|------|------|---|\n| 1 | Vocal separation | Demucs htdemucs (~9.2 dB SDR) |\n| 2 | Word-level transcription | Whisper large-v3 |\n| 3 | Song identification | AcoustID / LRCLIB text search |\n| 4 | Lyrics acquisition | LRCLIB (free, no auth) |\n| 5 | Alignment | Sequence alignment (LCS + fuzzy matching) |\n| 6 | Timing refinement | Onset/offset detection (librosa, 5.8ms resolution) |\n\n**Tips:** Provide artist & title for best results. Supported: MP3, WAV, FLAC, OGG, M4A. Processing: ~30-90s.")
if __name__ == "__main__":
demo.launch()