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