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import argparse |
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import sys |
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from pathlib import Path |
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import subprocess |
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import julius |
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import torch as th |
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import torchaudio as ta |
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from .audio import AudioFile, convert_audio_channels |
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from .pretrained import is_pretrained, load_pretrained |
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from .utils import apply_model, load_model |
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def load_track(track, device, audio_channels, samplerate): |
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errors = {} |
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wav = None |
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try: |
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wav = AudioFile(track).read( |
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streams=0, |
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samplerate=samplerate, |
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channels=audio_channels).to(device) |
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except FileNotFoundError: |
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errors['ffmpeg'] = 'Ffmpeg is not installed.' |
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except subprocess.CalledProcessError: |
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errors['ffmpeg'] = 'FFmpeg could not read the file.' |
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if wav is None: |
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try: |
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wav, sr = ta.load(str(track)) |
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except RuntimeError as err: |
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errors['torchaudio'] = err.args[0] |
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else: |
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wav = convert_audio_channels(wav, audio_channels) |
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wav = wav.to(device) |
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wav = julius.resample_frac(wav, sr, samplerate) |
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if wav is None: |
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print(f"Could not load file {track}. " |
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"Maybe it is not a supported file format? ") |
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for backend, error in errors.items(): |
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print(f"When trying to load using {backend}, got the following error: {error}") |
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sys.exit(1) |
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return wav |
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def encode_mp3(wav, path, bitrate=320, samplerate=44100, channels=2, verbose=False): |
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try: |
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import lameenc |
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except ImportError: |
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print("Failed to call lame encoder. Maybe it is not installed? " |
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"On windows, run `python.exe -m pip install -U lameenc`, " |
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"on OSX/Linux, run `python3 -m pip install -U lameenc`, " |
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"then try again.", file=sys.stderr) |
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sys.exit(1) |
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encoder = lameenc.Encoder() |
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encoder.set_bit_rate(bitrate) |
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encoder.set_in_sample_rate(samplerate) |
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encoder.set_channels(channels) |
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encoder.set_quality(2) |
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if not verbose: |
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encoder.silence() |
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wav = wav.transpose(0, 1).numpy() |
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mp3_data = encoder.encode(wav.tobytes()) |
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mp3_data += encoder.flush() |
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with open(path, "wb") as f: |
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f.write(mp3_data) |
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def main(): |
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parser = argparse.ArgumentParser("demucs.separate", |
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description="Separate the sources for the given tracks") |
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parser.add_argument("tracks", nargs='+', type=Path, default=[], help='Path to tracks') |
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parser.add_argument("-n", |
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"--name", |
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default="demucs_quantized", |
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help="Model name. See README.md for the list of pretrained models. " |
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"Default is demucs_quantized.") |
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parser.add_argument("-v", "--verbose", action="store_true") |
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parser.add_argument("-o", |
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"--out", |
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type=Path, |
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default=Path("separated"), |
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help="Folder where to put extracted tracks. A subfolder " |
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"with the model name will be created.") |
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parser.add_argument("--models", |
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type=Path, |
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default=Path("models"), |
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help="Path to trained models. " |
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"Also used to store downloaded pretrained models") |
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parser.add_argument("-d", |
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"--device", |
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default="cuda" if th.cuda.is_available() else "cpu", |
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help="Device to use, default is cuda if available else cpu") |
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parser.add_argument("--shifts", |
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default=0, |
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type=int, |
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help="Number of random shifts for equivariant stabilization." |
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"Increase separation time but improves quality for Demucs. 10 was used " |
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"in the original paper.") |
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parser.add_argument("--overlap", |
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default=0.25, |
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type=float, |
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help="Overlap between the splits.") |
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parser.add_argument("--no-split", |
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action="store_false", |
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dest="split", |
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default=True, |
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help="Doesn't split audio in chunks. This can use large amounts of memory.") |
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parser.add_argument("--float32", |
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action="store_true", |
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help="Convert the output wavefile to use pcm f32 format instead of s16. " |
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"This should not make a difference if you just plan on listening to the " |
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"audio but might be needed to compute exactly metrics like SDR etc.") |
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parser.add_argument("--int16", |
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action="store_false", |
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dest="float32", |
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help="Opposite of --float32, here for compatibility.") |
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parser.add_argument("--mp3", action="store_true", |
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help="Convert the output wavs to mp3.") |
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parser.add_argument("--mp3-bitrate", |
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default=320, |
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type=int, |
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help="Bitrate of converted mp3.") |
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args = parser.parse_args() |
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name = args.name + ".th" |
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model_path = args.models / name |
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if model_path.is_file(): |
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model = load_model(model_path) |
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else: |
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if is_pretrained(args.name): |
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model = load_pretrained(args.name) |
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else: |
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print(f"No pre-trained model {args.name}", file=sys.stderr) |
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sys.exit(1) |
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model.to(args.device) |
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out = args.out / args.name |
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out.mkdir(parents=True, exist_ok=True) |
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print(f"Separated tracks will be stored in {out.resolve()}") |
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for track in args.tracks: |
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if not track.exists(): |
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print( |
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f"File {track} does not exist. If the path contains spaces, " |
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"please try again after surrounding the entire path with quotes \"\".", |
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file=sys.stderr) |
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continue |
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print(f"Separating track {track}") |
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wav = load_track(track, args.device, model.audio_channels, model.samplerate) |
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ref = wav.mean(0) |
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wav = (wav - ref.mean()) / ref.std() |
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sources = apply_model(model, wav, shifts=args.shifts, split=args.split, |
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overlap=args.overlap, progress=True) |
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sources = sources * ref.std() + ref.mean() |
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track_folder = out / track.name.rsplit(".", 1)[0] |
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track_folder.mkdir(exist_ok=True) |
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for source, name in zip(sources, model.sources): |
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source = source / max(1.01 * source.abs().max(), 1) |
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if args.mp3 or not args.float32: |
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source = (source * 2**15).clamp_(-2**15, 2**15 - 1).short() |
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source = source.cpu() |
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stem = str(track_folder / name) |
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if args.mp3: |
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encode_mp3(source, stem + ".mp3", |
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bitrate=args.mp3_bitrate, |
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samplerate=model.samplerate, |
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channels=model.audio_channels, |
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verbose=args.verbose) |
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else: |
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wavname = str(track_folder / f"{name}.wav") |
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ta.save(wavname, source, sample_rate=model.samplerate) |
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if __name__ == "__main__": |
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main() |
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