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utils.py
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utils.py
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# coding: utf-8
__author__ = 'Roman Solovyev (ZFTurbo): https://github.com/ZFTurbo/'
import time
import numpy as np
import torch
import torch.nn as nn
import yaml
import librosa
import torch.nn.functional as F
from ml_collections import ConfigDict
from omegaconf import OmegaConf
from tqdm.auto import tqdm
from numpy.typing import NDArray
from typing import Dict, List
def get_model_from_config(model_type, config_path):
with open(config_path) as f:
if model_type == 'htdemucs':
config = OmegaConf.load(config_path)
else:
config = ConfigDict(yaml.load(f, Loader=yaml.FullLoader))
if model_type == 'mdx23c':
from models.mdx23c_tfc_tdf_v3 import TFC_TDF_net
model = TFC_TDF_net(config)
elif model_type == 'htdemucs':
from models.demucs4ht import get_model
model = get_model(config)
elif model_type == 'segm_models':
from models.segm_models import Segm_Models_Net
model = Segm_Models_Net(config)
elif model_type == 'torchseg':
from models.torchseg_models import Torchseg_Net
model = Torchseg_Net(config)
elif model_type == 'mel_band_roformer':
from models.bs_roformer import MelBandRoformer
model = MelBandRoformer(
**dict(config.model)
)
elif model_type == 'bs_roformer':
from models.bs_roformer import BSRoformer
model = BSRoformer(
**dict(config.model)
)
elif model_type == 'swin_upernet':
from models.upernet_swin_transformers import Swin_UperNet_Model
model = Swin_UperNet_Model(config)
elif model_type == 'bandit':
from models.bandit.core.model import MultiMaskMultiSourceBandSplitRNNSimple
model = MultiMaskMultiSourceBandSplitRNNSimple(
**config.model
)
elif model_type == 'bandit_v2':
from models.bandit_v2.bandit import Bandit
model = Bandit(
**config.kwargs
)
elif model_type == 'scnet_unofficial':
from models.scnet_unofficial import SCNet
model = SCNet(
**config.model
)
elif model_type == 'scnet':
from models.scnet import SCNet
model = SCNet(
**config.model
)
elif model_type == 'apollo':
from models.look2hear.models import BaseModel
model = BaseModel.apollo(**config.model)
elif model_type == 'bs_mamba2':
from models.ts_bs_mamba2 import Separator
model = Separator(**config.model)
else:
print('Unknown model: {}'.format(model_type))
model = None
return model, config
def _getWindowingArray(window_size, fade_size):
fadein = torch.linspace(0, 1, fade_size)
fadeout = torch.linspace(1, 0, fade_size)
window = torch.ones(window_size)
window[-fade_size:] *= fadeout
window[:fade_size] *= fadein
return window
def demix_track(config, model, mix, device, pbar=False):
C = config.audio.chunk_size
N = config.inference.num_overlap
fade_size = C // 10
step = int(C // N)
border = C - step
batch_size = config.inference.batch_size
length_init = mix.shape[-1]
# Do pad from the beginning and end to account floating window results better
if length_init > 2 * border and (border > 0):
mix = nn.functional.pad(mix, (border, border), mode='reflect')
# windowingArray crossfades at segment boundaries to mitigate clicking artifacts
windowingArray = _getWindowingArray(C, fade_size)
with torch.cuda.amp.autocast(enabled=config.training.use_amp):
with torch.inference_mode():
req_shape = (len(prefer_target_instrument(config)),) + tuple(mix.shape)
result = torch.zeros(req_shape, dtype=torch.float32)
counter = torch.zeros(req_shape, dtype=torch.float32)
i = 0
batch_data = []
batch_locations = []
progress_bar = tqdm(total=mix.shape[1], desc="Processing audio chunks", leave=False) if pbar else None
while i < mix.shape[1]:
# print(i, i + C, mix.shape[1])
part = mix[:, i:i + C].to(device)
length = part.shape[-1]
if length < C:
if length > C // 2 + 1:
part = nn.functional.pad(input=part, pad=(0, C - length), mode='reflect')
else:
part = nn.functional.pad(input=part, pad=(0, C - length, 0, 0), mode='constant', value=0)
batch_data.append(part)
batch_locations.append((i, length))
i += step
if len(batch_data) >= batch_size or (i >= mix.shape[1]):
arr = torch.stack(batch_data, dim=0)
x = model(arr)
window = windowingArray
if i - step == 0: # First audio chunk, no fadein
window[:fade_size] = 1
elif i >= mix.shape[1]: # Last audio chunk, no fadeout
window[-fade_size:] = 1
for j in range(len(batch_locations)):
start, l = batch_locations[j]
result[..., start:start+l] += x[j][..., :l].cpu() * window[..., :l]
counter[..., start:start+l] += window[..., :l]
batch_data = []
batch_locations = []
if progress_bar:
progress_bar.update(step)
if progress_bar:
progress_bar.close()
estimated_sources = result / counter
estimated_sources = estimated_sources.cpu().numpy()
np.nan_to_num(estimated_sources, copy=False, nan=0.0)
if length_init > 2 * border and (border > 0):
# Remove pad
estimated_sources = estimated_sources[..., border:-border]
return {k: v for k, v in zip(prefer_target_instrument(config), estimated_sources)}
def demix_track_demucs(config, model, mix, device, pbar=False):
S = len(config.training.instruments)
C = config.training.samplerate * config.training.segment
N = config.inference.num_overlap
batch_size = config.inference.batch_size
step = C // N
# print(S, C, N, step, mix.shape, mix.device)
with torch.cuda.amp.autocast(enabled=config.training.use_amp):
with torch.inference_mode():
req_shape = (S, ) + tuple(mix.shape)
result = torch.zeros(req_shape, dtype=torch.float32)
counter = torch.zeros(req_shape, dtype=torch.float32)
i = 0
batch_data = []
batch_locations = []
progress_bar = tqdm(total=mix.shape[1], desc="Processing audio chunks", leave=False) if pbar else None
while i < mix.shape[1]:
# print(i, i + C, mix.shape[1])
part = mix[:, i:i + C].to(device)
length = part.shape[-1]
if length < C:
part = nn.functional.pad(input=part, pad=(0, C - length, 0, 0), mode='constant', value=0)
batch_data.append(part)
batch_locations.append((i, length))
i += step
if len(batch_data) >= batch_size or (i >= mix.shape[1]):
arr = torch.stack(batch_data, dim=0)
x = model(arr)
for j in range(len(batch_locations)):
start, l = batch_locations[j]
result[..., start:start+l] += x[j][..., :l].cpu()
counter[..., start:start+l] += 1.
batch_data = []
batch_locations = []
if progress_bar:
progress_bar.update(step)
if progress_bar:
progress_bar.close()
estimated_sources = result / counter
estimated_sources = estimated_sources.cpu().numpy()
np.nan_to_num(estimated_sources, copy=False, nan=0.0)
if S > 1:
return {k: v for k, v in zip(config.training.instruments, estimated_sources)}
else:
return estimated_sources
def sdr(references, estimates):
# compute SDR for one song
delta = 1e-7 # avoid numerical errors
num = np.sum(np.square(references), axis=(1, 2))
den = np.sum(np.square(references - estimates), axis=(1, 2))
num += delta
den += delta
return 10 * np.log10(num / den)
def si_sdr(reference, estimate):
eps = 1e-07
scale = np.sum(estimate * reference + eps, axis=(0, 1)) / np.sum(reference**2 + eps, axis=(0, 1))
scale = np.expand_dims(scale, axis=(0, 1)) # shape - [50, 1]
reference = reference * scale
sisdr = np.mean(10 * np.log10(np.sum(reference**2, axis=(0, 1)) / (np.sum((reference - estimate)**2, axis=(0, 1)) + eps) + eps))
return sisdr
def L1Freq_metric(
reference,
estimate,
fft_size=2048,
hop_size=1024,
device='cpu'
):
reference = torch.from_numpy(reference).to(device)
estimate = torch.from_numpy(estimate).to(device)
reference_stft = torch.stft(reference, fft_size, hop_size, return_complex=True)
estimated_stft = torch.stft(estimate, fft_size, hop_size, return_complex=True)
reference_mag = torch.abs(reference_stft)
estimate_mag = torch.abs(estimated_stft)
loss = 10 * F.l1_loss(estimate_mag, reference_mag)
# Metric is on the range from 0 to 100 - larger the better
ret = 100 / (1. + float(loss.cpu().numpy()))
return ret
def LogWMSE_metric(
reference,
estimate,
mixture,
device='cpu',
):
from torch_log_wmse import LogWMSE
log_wmse = LogWMSE(
audio_length=reference.shape[-1] / 44100,
sample_rate=44100,
return_as_loss=False, # optional
bypass_filter=False, # optional
)
reference = torch.from_numpy(reference).unsqueeze(0).unsqueeze(0).to(device)
estimate = torch.from_numpy(estimate).unsqueeze(0).unsqueeze(0).to(device)
mixture = torch.from_numpy(mixture).unsqueeze(0).to(device)
# print(reference.shape, estimate.shape, mixture.shape)
res = log_wmse(mixture, reference, estimate)
return float(res.cpu().numpy())
def AuraSTFT_metric(
reference,
estimate,
device='cpu',
):
from auraloss.freq import STFTLoss
stft_loss = STFTLoss(
w_log_mag=1.0,
w_lin_mag=0.0,
w_sc=1.0,
device=device,
)
reference = torch.from_numpy(reference).unsqueeze(0).to(device)
estimate = torch.from_numpy(estimate).unsqueeze(0).to(device)
res = 100 / (1. + 10 * stft_loss(reference, estimate))
return float(res.cpu().numpy())
def AuraMRSTFT_metric(
reference,
estimate,
device='cpu',
):
from auraloss.freq import MultiResolutionSTFTLoss
mrstft_loss = MultiResolutionSTFTLoss(
fft_sizes=[1024, 2048, 4096],
hop_sizes=[256, 512, 1024],
win_lengths=[1024, 2048, 4096],
scale="mel",
n_bins=128,
sample_rate=44100,
perceptual_weighting=True,
device=device
)
reference = torch.from_numpy(reference).unsqueeze(0).float().to(device)
estimate = torch.from_numpy(estimate).unsqueeze(0).float().to(device)
res = 100 / (1. + 10 * mrstft_loss(reference, estimate))
return float(res.cpu().numpy())
def bleed_full(
reference,
estimate,
sr=44100,
n_fft=4096,
hop_length=1024,
n_mels=512,
device='cpu',
):
from torchaudio.transforms import AmplitudeToDB
# Move tensors to GPU if available
reference = torch.from_numpy(reference).float().to(device)
estimate = torch.from_numpy(estimate).float().to(device)
# Create a Hann window
window = torch.hann_window(n_fft).to(device)
# Compute STFTs with the Hann window
D1 = torch.abs(torch.stft(reference, n_fft=n_fft, hop_length=hop_length, window=window, return_complex=True, pad_mode="constant"))
D2 = torch.abs(torch.stft(estimate, n_fft=n_fft, hop_length=hop_length, window=window, return_complex=True, pad_mode="constant"))
# create mel filterbank
mel_basis = librosa.filters.mel(sr=sr, n_fft=n_fft, n_mels=n_mels)
mel_filter_bank = torch.from_numpy(mel_basis).to(device) # (melbandroformer is doing it that way) edit: sent to right device now
# apply mel scale
S1_mel = torch.matmul(mel_filter_bank, D1)
S2_mel = torch.matmul(mel_filter_bank, D2)
# Convert to decibels
S1_db = AmplitudeToDB(stype="magnitude", top_db=80)(S1_mel)
S2_db = AmplitudeToDB(stype="magnitude", top_db=80)(S2_mel)
# Calculate difference
diff = S2_db - S1_db
# Separate positive and negative differences
positive_diff = diff[diff > 0]
negative_diff = diff[diff < 0]
# Calculate averages
average_positive = torch.mean(positive_diff) if positive_diff.numel() > 0 else torch.tensor(0.0).to(device)
average_negative = torch.mean(negative_diff) if negative_diff.numel() > 0 else torch.tensor(0.0).to(device)
# Scale with 100 as best score
bleedless = 100 * 1 / (average_positive + 1)
fullness = 100 * 1 / (-average_negative + 1)
return bleedless.cpu().numpy(), fullness.cpu().numpy()
def get_metrics(
metrics,
reference, # (ch, length)
estimate, # (ch, length)
mix, # (ch, length)
device='cpu',
):
result = dict()
if 'sdr' in metrics:
references = np.expand_dims(reference, axis=0)
estimates = np.expand_dims(estimate, axis=0)
result['sdr'] = sdr(references, estimates)[0]
if 'si_sdr' in metrics:
result['si_sdr'] = si_sdr(reference, estimate)
if 'l1_freq' in metrics:
result['l1_freq'] = L1Freq_metric(reference, estimate, device=device)
if 'log_wmse' in metrics:
result['log_wmse'] = LogWMSE_metric(reference, estimate, mix, device)
if 'aura_stft' in metrics:
result['aura_stft'] = AuraSTFT_metric(reference, estimate, device)
if 'aura_mrstft' in metrics:
result['aura_mrstft'] = AuraMRSTFT_metric(reference, estimate, device)
if 'bleedless' in metrics or 'fullness' in metrics:
bleedless, fullness = bleed_full(reference, estimate, device=device)
if 'bleedless' in metrics:
result['bleedless'] = bleedless
if 'fullness' in metrics:
result['fullness'] = fullness
return result
def demix(config, model, mix: NDArray, device, pbar=False, model_type: str = None) -> Dict[str, NDArray]:
mix = torch.tensor(mix, dtype=torch.float32)
if model_type == 'htdemucs':
return demix_track_demucs(config, model, mix, device, pbar=pbar)
else:
return demix_track(config, model, mix, device, pbar=pbar)
def prefer_target_instrument(config: ConfigDict) -> List[str]:
if config.training.get('target_instrument'):
return [config.training.target_instrument]
else:
return config.training.instruments