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dfd1909 | 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 | #!/usr/bin/env python3
"""
Audio super-resolution using FlashSR.
Independently written wrapper around the FlashSR model by Jaekwon Im and
Juhan Nam (KAIST). Supports files of arbitrary length via windowed processing
with overlap-add. No dependency on torchcodec or FFmpeg -- uses soundfile for
all I/O.
Paper: https://arxiv.org/abs/2501.10807
"""
from __future__ import annotations
import argparse
import math
import os
import sys
import time
from pathlib import Path
import numpy as np
import soundfile as sf
import torch
from scipy.signal import resample_poly
from FlashSR.FlashSR import FlashSR
# ---- constants ----------------------------------------------------------------
TARGET_SR = 48_000
WINDOW_LEN = 245_760 # samples per model call (5.12 s at 48 kHz)
OVERLAP = 24_000 # crossfade region (0.50 s)
HOP = WINDOW_LEN - OVERLAP # advance per window (4.62 s)
AUDIO_EXTENSIONS = {".wav", ".flac", ".mp3", ".ogg", ".opus"}
# ---- helpers ------------------------------------------------------------------
def _load_mono(path: str | Path) -> tuple[np.ndarray, int]:
"""Read an audio file, mix to mono, return (float32 array, sample_rate)."""
data, sr = sf.read(str(path), dtype="float32")
if data.ndim == 2:
data = data.mean(axis=1)
return data, sr
def _resample_if_needed(audio: np.ndarray, orig_sr: int) -> np.ndarray:
"""Polyphase resample to TARGET_SR when the source rate differs."""
if orig_sr == TARGET_SR:
return audio
return resample_poly(audio, TARGET_SR, orig_sr).astype(np.float32)
def _build_fade(length: int) -> torch.Tensor:
"""Half-cosine fade-in ramp of *length* samples (0 -> 1)."""
t = torch.linspace(0.0, math.pi / 2, length)
return torch.sin(t) ** 2 # cos^2 fade is smooth at both ends
def _pad_to(tensor: torch.Tensor, n: int) -> torch.Tensor:
"""Right-zero-pad the last dimension to at least *n* samples."""
deficit = n - tensor.shape[-1]
if deficit <= 0:
return tensor
return torch.nn.functional.pad(tensor, (0, deficit))
# ---- core ---------------------------------------------------------------------
def build_model(weights_dir: str | Path, device: torch.device) -> FlashSR:
"""Instantiate FlashSR and load pretrained weights."""
w = Path(weights_dir)
model = FlashSR(
student_ldm_ckpt_path=str(w / "student_ldm.pth"),
sr_vocoder_ckpt_path=str(w / "sr_vocoder.pth"),
autoencoder_ckpt_path=str(w / "vae.pth"),
)
return model.to(device).eval()
@torch.inference_mode()
def enhance(
model: FlashSR,
waveform: np.ndarray,
*,
device: torch.device,
lowpass: bool = False,
) -> np.ndarray:
"""
Run FlashSR on a mono waveform (numpy float32, 48 kHz).
Long inputs are split into overlapping windows and reassembled with
overlap-add using a raised-cosine crossfade.
Returns enhanced waveform as numpy float32 at 48 kHz.
"""
signal = torch.from_numpy(waveform).unsqueeze(0) # (1, T)
n_samples = signal.shape[-1]
# --- short signal: single pass -------------------------------------------
if n_samples <= WINDOW_LEN:
chunk = _pad_to(signal, WINDOW_LEN).to(device)
out = model(chunk, lowpass_input=lowpass)
return out[0, :n_samples].cpu().numpy()
# --- long signal: overlap-add --------------------------------------------
fade = _build_fade(OVERLAP)
accumulator = torch.zeros(n_samples)
norm = torch.zeros(n_samples)
offset = 0
while offset < n_samples:
end = min(offset + WINDOW_LEN, n_samples)
segment = signal[:, offset:end]
segment = _pad_to(segment, WINDOW_LEN).to(device)
enhanced_seg = model(segment, lowpass_input=lowpass).cpu().squeeze(0)
seg_len = min(WINDOW_LEN, n_samples - offset)
enhanced_seg = enhanced_seg[:seg_len]
# per-sample weights: 1.0 everywhere, faded in at overlap boundary
w = torch.ones(seg_len)
if offset > 0 and seg_len > OVERLAP:
w[:OVERLAP] = fade
accumulator[offset : offset + seg_len] += enhanced_seg * w
norm[offset : offset + seg_len] += w
offset += HOP
norm.clamp_(min=1e-8)
return (accumulator / norm).numpy()
# ---- file-level convenience ---------------------------------------------------
def enhance_file(
model: FlashSR,
src: str | Path,
dst: str | Path,
*,
device: torch.device,
lowpass: bool = False,
) -> float:
"""Enhance one file. Returns duration in seconds."""
raw, sr = _load_mono(src)
audio = _resample_if_needed(raw, sr)
result = enhance(model, audio, device=device, lowpass=lowpass)
os.makedirs(os.path.dirname(dst) or ".", exist_ok=True)
sf.write(str(dst), result, TARGET_SR)
return len(audio) / TARGET_SR
def collect_audio_files(root: str | Path) -> list[Path]:
"""Recursively find audio files under *root*."""
root = Path(root)
return sorted(p for p in root.rglob("*") if p.suffix.lower() in AUDIO_EXTENSIONS)
# ---- CLI ----------------------------------------------------------------------
def cli() -> None:
ap = argparse.ArgumentParser(
description="FlashSR audio super-resolution (by Im & Nam, KAIST)")
ap.add_argument("--input", "-i", required=True,
help="Input audio file or directory")
ap.add_argument("--output", "-o", required=True,
help="Output file or directory")
ap.add_argument("--weights", "-w", default="./weights",
help="Directory containing the three .pth weight files")
ap.add_argument("--lowpass", action="store_true",
help="Apply lowpass filter before enhancement")
ap.add_argument("--device", default="cuda",
help="Torch device (default: cuda)")
args = ap.parse_args()
dev = torch.device(args.device if torch.cuda.is_available() else "cpu")
print(f"Device: {dev}")
print("Loading model...")
t0 = time.monotonic()
model = build_model(args.weights, dev)
print(f"Loaded in {time.monotonic() - t0:.1f}s")
# Resolve inputs
inp = Path(args.input)
out = Path(args.output)
if inp.is_dir():
files = collect_audio_files(inp)
if not files:
sys.exit(f"No audio files found in {inp}")
pairs = [(f, out / f.relative_to(inp)) for f in files]
else:
pairs = [(inp, out)]
total_dur = 0.0
t_start = time.monotonic()
for idx, (src, dst) in enumerate(pairs, 1):
print(f"[{idx}/{len(pairs)}] {src} -> {dst}")
dur = enhance_file(model, src, dst, device=dev, lowpass=args.lowpass)
total_dur += dur
print(f" {dur:.1f}s of audio")
elapsed = time.monotonic() - t_start
rtf = total_dur / elapsed if elapsed > 0 else 0
print(f"\nDone: {len(pairs)} file(s), {total_dur:.1f}s audio, "
f"{elapsed:.1f}s wall-clock ({rtf:.1f}x realtime)")
if __name__ == "__main__":
cli()
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