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import os | |
import datetime | |
import json | |
import logging | |
import librosa | |
import pickle | |
from typing import Dict | |
import numpy as np | |
import torch | |
import torch.nn as nn | |
import yaml | |
from models.audiosep import AudioSep, get_model_class | |
def ignore_warnings(): | |
import warnings | |
# Ignore UserWarning from torch.meshgrid | |
warnings.filterwarnings('ignore', category=UserWarning, module='torch.functional') | |
# Refined regex pattern to capture variations in the warning message | |
pattern = r"Some weights of the model checkpoint at roberta-base were not used when initializing RobertaModel: \['lm_head\..*'\].*" | |
warnings.filterwarnings('ignore', message=pattern) | |
def create_logging(log_dir, filemode): | |
os.makedirs(log_dir, exist_ok=True) | |
i1 = 0 | |
while os.path.isfile(os.path.join(log_dir, "{:04d}.log".format(i1))): | |
i1 += 1 | |
log_path = os.path.join(log_dir, "{:04d}.log".format(i1)) | |
logging.basicConfig( | |
level=logging.DEBUG, | |
format="%(asctime)s %(filename)s[line:%(lineno)d] %(levelname)s %(message)s", | |
datefmt="%a, %d %b %Y %H:%M:%S", | |
filename=log_path, | |
filemode=filemode, | |
) | |
# Print to console | |
console = logging.StreamHandler() | |
console.setLevel(logging.INFO) | |
formatter = logging.Formatter("%(name)-12s: %(levelname)-8s %(message)s") | |
console.setFormatter(formatter) | |
logging.getLogger("").addHandler(console) | |
return logging | |
def float32_to_int16(x: float) -> int: | |
x = np.clip(x, a_min=-1, a_max=1) | |
return (x * 32767.0).astype(np.int16) | |
def int16_to_float32(x: int) -> float: | |
return (x / 32767.0).astype(np.float32) | |
def parse_yaml(config_yaml: str) -> Dict: | |
r"""Parse yaml file. | |
Args: | |
config_yaml (str): config yaml path | |
Returns: | |
yaml_dict (Dict): parsed yaml file | |
""" | |
with open(config_yaml, "r") as fr: | |
return yaml.load(fr, Loader=yaml.FullLoader) | |
def get_audioset632_id_to_lb(ontology_path: str) -> Dict: | |
r"""Get AudioSet 632 classes ID to label mapping.""" | |
audioset632_id_to_lb = {} | |
with open(ontology_path) as f: | |
data_list = json.load(f) | |
for e in data_list: | |
audioset632_id_to_lb[e["id"]] = e["name"] | |
return audioset632_id_to_lb | |
def load_pretrained_panns( | |
model_type: str, | |
checkpoint_path: str, | |
freeze: bool | |
) -> nn.Module: | |
r"""Load pretrained pretrained audio neural networks (PANNs). | |
Args: | |
model_type: str, e.g., "Cnn14" | |
checkpoint_path, str, e.g., "Cnn14_mAP=0.431.pth" | |
freeze: bool | |
Returns: | |
model: nn.Module | |
""" | |
if model_type == "Cnn14": | |
Model = Cnn14 | |
elif model_type == "Cnn14_DecisionLevelMax": | |
Model = Cnn14_DecisionLevelMax | |
else: | |
raise NotImplementedError | |
model = Model(sample_rate=32000, window_size=1024, hop_size=320, | |
mel_bins=64, fmin=50, fmax=14000, classes_num=527) | |
if checkpoint_path: | |
checkpoint = torch.load(checkpoint_path, map_location="cpu") | |
model.load_state_dict(checkpoint["model"]) | |
if freeze: | |
for param in model.parameters(): | |
param.requires_grad = False | |
return model | |
def energy(x): | |
return torch.mean(x ** 2) | |
def magnitude_to_db(x): | |
eps = 1e-10 | |
return 20. * np.log10(max(x, eps)) | |
def db_to_magnitude(x): | |
return 10. ** (x / 20) | |
def ids_to_hots(ids, classes_num, device): | |
hots = torch.zeros(classes_num).to(device) | |
for id in ids: | |
hots[id] = 1 | |
return hots | |
def calculate_sdr( | |
ref: np.ndarray, | |
est: np.ndarray, | |
eps=1e-10 | |
) -> float: | |
r"""Calculate SDR between reference and estimation. | |
Args: | |
ref (np.ndarray), reference signal | |
est (np.ndarray), estimated signal | |
""" | |
reference = ref | |
noise = est - reference | |
numerator = np.clip(a=np.mean(reference ** 2), a_min=eps, a_max=None) | |
denominator = np.clip(a=np.mean(noise ** 2), a_min=eps, a_max=None) | |
sdr = 10. * np.log10(numerator / denominator) | |
return sdr | |
def calculate_sisdr(ref, est): | |
r"""Calculate SDR between reference and estimation. | |
Args: | |
ref (np.ndarray), reference signal | |
est (np.ndarray), estimated signal | |
""" | |
eps = np.finfo(ref.dtype).eps | |
reference = ref.copy() | |
estimate = est.copy() | |
reference = reference.reshape(reference.size, 1) | |
estimate = estimate.reshape(estimate.size, 1) | |
Rss = np.dot(reference.T, reference) | |
# get the scaling factor for clean sources | |
a = (eps + np.dot(reference.T, estimate)) / (Rss + eps) | |
e_true = a * reference | |
e_res = estimate - e_true | |
Sss = (e_true**2).sum() | |
Snn = (e_res**2).sum() | |
sisdr = 10 * np.log10((eps+ Sss)/(eps + Snn)) | |
return sisdr | |
class StatisticsContainer(object): | |
def __init__(self, statistics_path): | |
self.statistics_path = statistics_path | |
self.backup_statistics_path = "{}_{}.pkl".format( | |
os.path.splitext(self.statistics_path)[0], | |
datetime.datetime.now().strftime("%Y-%m-%d_%H-%M-%S"), | |
) | |
self.statistics_dict = {"balanced_train": [], "test": []} | |
def append(self, steps, statistics, split, flush=True): | |
statistics["steps"] = steps | |
self.statistics_dict[split].append(statistics) | |
if flush: | |
self.flush() | |
def flush(self): | |
pickle.dump(self.statistics_dict, open(self.statistics_path, "wb")) | |
pickle.dump(self.statistics_dict, open(self.backup_statistics_path, "wb")) | |
logging.info(" Dump statistics to {}".format(self.statistics_path)) | |
logging.info(" Dump statistics to {}".format(self.backup_statistics_path)) | |
def get_mean_sdr_from_dict(sdris_dict): | |
mean_sdr = np.nanmean(list(sdris_dict.values())) | |
return mean_sdr | |
def remove_silence(audio: np.ndarray, sample_rate: int) -> np.ndarray: | |
r"""Remove silent frames.""" | |
window_size = int(sample_rate * 0.1) | |
threshold = 0.02 | |
frames = librosa.util.frame(x=audio, frame_length=window_size, hop_length=window_size).T | |
# shape: (frames_num, window_size) | |
new_frames = get_active_frames(frames, threshold) | |
# shape: (new_frames_num, window_size) | |
new_audio = new_frames.flatten() | |
# shape: (new_audio_samples,) | |
return new_audio | |
def get_active_frames(frames: np.ndarray, threshold: float) -> np.ndarray: | |
r"""Get active frames.""" | |
energy = np.max(np.abs(frames), axis=-1) | |
# shape: (frames_num,) | |
active_indexes = np.where(energy > threshold)[0] | |
# shape: (new_frames_num,) | |
new_frames = frames[active_indexes] | |
# shape: (new_frames_num,) | |
return new_frames | |
def repeat_to_length(audio: np.ndarray, segment_samples: int) -> np.ndarray: | |
r"""Repeat audio to length.""" | |
repeats_num = (segment_samples // audio.shape[-1]) + 1 | |
audio = np.tile(audio, repeats_num)[0 : segment_samples] | |
return audio | |
def calculate_segmentwise_sdr(ref, est, hop_samples, return_sdr_list=False): | |
min_len = min(ref.shape[-1], est.shape[-1]) | |
pointer = 0 | |
sdrs = [] | |
while pointer + hop_samples < min_len: | |
sdr = calculate_sdr( | |
ref=ref[:, pointer : pointer + hop_samples], | |
est=est[:, pointer : pointer + hop_samples], | |
) | |
sdrs.append(sdr) | |
pointer += hop_samples | |
sdr = np.nanmedian(sdrs) | |
if return_sdr_list: | |
return sdr, sdrs | |
else: | |
return sdr | |
def loudness(data, input_loudness, target_loudness): | |
""" Loudness normalize a signal. | |
Normalize an input signal to a user loudness in dB LKFS. | |
Params | |
------- | |
data : torch.Tensor | |
Input multichannel audio data. | |
input_loudness : float | |
Loudness of the input in dB LUFS. | |
target_loudness : float | |
Target loudness of the output in dB LUFS. | |
Returns | |
------- | |
output : torch.Tensor | |
Loudness normalized output data. | |
""" | |
# calculate the gain needed to scale to the desired loudness level | |
delta_loudness = target_loudness - input_loudness | |
gain = torch.pow(10.0, delta_loudness / 20.0) | |
output = gain * data | |
# check for potentially clipped samples | |
# if torch.max(torch.abs(output)) >= 1.0: | |
# warnings.warn("Possible clipped samples in output.") | |
return output | |
def load_ss_model( | |
configs: Dict, | |
checkpoint_path: str, | |
query_encoder: nn.Module | |
) -> nn.Module: | |
r"""Load trained universal source separation model. | |
Args: | |
configs (Dict) | |
checkpoint_path (str): path of the checkpoint to load | |
device (str): e.g., "cpu" | "cuda" | |
Returns: | |
pl_model: pl.LightningModule | |
""" | |
ss_model_type = configs["model"]["model_type"] | |
input_channels = configs["model"]["input_channels"] | |
output_channels = configs["model"]["output_channels"] | |
condition_size = configs["model"]["condition_size"] | |
# Initialize separation model | |
SsModel = get_model_class(model_type=ss_model_type) | |
ss_model = SsModel( | |
input_channels=input_channels, | |
output_channels=output_channels, | |
condition_size=condition_size, | |
) | |
# Load PyTorch Lightning model | |
pl_model = AudioSep.load_from_checkpoint( | |
checkpoint_path=checkpoint_path, | |
strict=False, | |
ss_model=ss_model, | |
waveform_mixer=None, | |
query_encoder=query_encoder, | |
loss_function=None, | |
optimizer_type=None, | |
learning_rate=None, | |
lr_lambda_func=None, | |
map_location=torch.device('cpu'), | |
) | |
return pl_model | |
def parse_yaml(config_yaml: str) -> Dict: | |
r"""Parse yaml file. | |
Args: | |
config_yaml (str): config yaml path | |
Returns: | |
yaml_dict (Dict): parsed yaml file | |
""" | |
with open(config_yaml, "r") as fr: | |
return yaml.load(fr, Loader=yaml.FullLoader) |