conex / espnet2 /bin /enh_inference.py
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#!/usr/bin/env python3
import argparse
import logging
from pathlib import Path
import sys
from typing import List
from typing import Optional
from typing import Sequence
from typing import Tuple
from typing import Union
import humanfriendly
import numpy as np
import torch
from tqdm import trange
from typeguard import check_argument_types
from espnet.utils.cli_utils import get_commandline_args
from espnet2.fileio.sound_scp import SoundScpWriter
from espnet2.tasks.enh import EnhancementTask
from espnet2.torch_utils.device_funcs import to_device
from espnet2.torch_utils.set_all_random_seed import set_all_random_seed
from espnet2.utils import config_argparse
from espnet2.utils.types import str2bool
from espnet2.utils.types import str2triple_str
from espnet2.utils.types import str_or_none
EPS = torch.finfo(torch.get_default_dtype()).eps
class SeparateSpeech:
"""SeparateSpeech class
Examples:
>>> import soundfile
>>> separate_speech = SeparateSpeech("enh_config.yml", "enh.pth")
>>> audio, rate = soundfile.read("speech.wav")
>>> separate_speech(audio)
[separated_audio1, separated_audio2, ...]
"""
def __init__(
self,
enh_train_config: Union[Path, str],
enh_model_file: Union[Path, str] = None,
segment_size: Optional[float] = None,
hop_size: Optional[float] = None,
normalize_segment_scale: bool = False,
show_progressbar: bool = False,
ref_channel: Optional[int] = None,
normalize_output_wav: bool = False,
device: str = "cpu",
dtype: str = "float32",
):
assert check_argument_types()
# 1. Build Enh model
enh_model, enh_train_args = EnhancementTask.build_model_from_file(
enh_train_config, enh_model_file, device
)
enh_model.to(dtype=getattr(torch, dtype)).eval()
self.device = device
self.dtype = dtype
self.enh_train_args = enh_train_args
self.enh_model = enh_model
# only used when processing long speech, i.e.
# segment_size is not None and hop_size is not None
self.segment_size = segment_size
self.hop_size = hop_size
self.normalize_segment_scale = normalize_segment_scale
self.normalize_output_wav = normalize_output_wav
self.show_progressbar = show_progressbar
self.num_spk = enh_model.num_spk
task = "enhancement" if self.num_spk == 1 else "separation"
# reference channel for processing multi-channel speech
if ref_channel is not None:
logging.info(
"Overwrite enh_model.separator.ref_channel with {}".format(ref_channel)
)
enh_model.separator.ref_channel = ref_channel
self.ref_channel = ref_channel
else:
self.ref_channel = enh_model.ref_channel
self.segmenting = segment_size is not None and hop_size is not None
if self.segmenting:
logging.info("Perform segment-wise speech %s" % task)
logging.info(
"Segment length = {} sec, hop length = {} sec".format(
segment_size, hop_size
)
)
else:
logging.info("Perform direct speech %s on the input" % task)
@torch.no_grad()
def __call__(
self, speech_mix: Union[torch.Tensor, np.ndarray], fs: int = 8000
) -> List[torch.Tensor]:
"""Inference
Args:
speech_mix: Input speech data (Batch, Nsamples [, Channels])
fs: sample rate
Returns:
[separated_audio1, separated_audio2, ...]
"""
assert check_argument_types()
# Input as audio signal
if isinstance(speech_mix, np.ndarray):
speech_mix = torch.as_tensor(speech_mix)
assert speech_mix.dim() > 1, speech_mix.size()
batch_size = speech_mix.size(0)
speech_mix = speech_mix.to(getattr(torch, self.dtype))
# lenghts: (B,)
lengths = speech_mix.new_full(
[batch_size], dtype=torch.long, fill_value=speech_mix.size(1)
)
# a. To device
speech_mix = to_device(speech_mix, device=self.device)
lengths = to_device(lengths, device=self.device)
if self.segmenting and lengths[0] > self.segment_size * fs:
# Segment-wise speech enhancement/separation
overlap_length = int(np.round(fs * (self.segment_size - self.hop_size)))
num_segments = int(
np.ceil((speech_mix.size(1) - overlap_length) / (self.hop_size * fs))
)
t = T = int(self.segment_size * fs)
pad_shape = speech_mix[:, :T].shape
enh_waves = []
range_ = trange if self.show_progressbar else range
for i in range_(num_segments):
st = int(i * self.hop_size * fs)
en = st + T
if en >= lengths[0]:
# en - st < T (last segment)
en = lengths[0]
speech_seg = speech_mix.new_zeros(pad_shape)
t = en - st
speech_seg[:, :t] = speech_mix[:, st:en]
else:
t = T
speech_seg = speech_mix[:, st:en] # B x T [x C]
lengths_seg = speech_mix.new_full(
[batch_size], dtype=torch.long, fill_value=T
)
# b. Enhancement/Separation Forward
feats, f_lens = self.enh_model.encoder(speech_seg, lengths_seg)
feats, _, _ = self.enh_model.separator(feats, f_lens)
processed_wav = [
self.enh_model.decoder(f, lengths_seg)[0] for f in feats
]
if speech_seg.dim() > 2:
# multi-channel speech
speech_seg_ = speech_seg[:, self.ref_channel]
else:
speech_seg_ = speech_seg
if self.normalize_segment_scale:
# normalize the energy of each separated stream
# to match the input energy
processed_wav = [
self.normalize_scale(w, speech_seg_) for w in processed_wav
]
# List[torch.Tensor(num_spk, B, T)]
enh_waves.append(torch.stack(processed_wav, dim=0))
# c. Stitch the enhanced segments together
waves = enh_waves[0]
for i in range(1, num_segments):
# permutation between separated streams in last and current segments
perm = self.cal_permumation(
waves[:, :, -overlap_length:],
enh_waves[i][:, :, :overlap_length],
criterion="si_snr",
)
# repermute separated streams in current segment
for batch in range(batch_size):
enh_waves[i][:, batch] = enh_waves[i][perm[batch], batch]
if i == num_segments - 1:
enh_waves[i][:, :, t:] = 0
enh_waves_res_i = enh_waves[i][:, :, overlap_length:t]
else:
enh_waves_res_i = enh_waves[i][:, :, overlap_length:]
# overlap-and-add (average over the overlapped part)
waves[:, :, -overlap_length:] = (
waves[:, :, -overlap_length:] + enh_waves[i][:, :, :overlap_length]
) / 2
# concatenate the residual parts of the later segment
waves = torch.cat([waves, enh_waves_res_i], dim=2)
# ensure the stitched length is same as input
assert waves.size(2) == speech_mix.size(1), (waves.shape, speech_mix.shape)
waves = torch.unbind(waves, dim=0)
else:
# b. Enhancement/Separation Forward
feats, f_lens = self.enh_model.encoder(speech_mix, lengths)
feats, _, _ = self.enh_model.separator(feats, f_lens)
waves = [self.enh_model.decoder(f, lengths)[0] for f in feats]
assert len(waves) == self.num_spk, len(waves) == self.num_spk
assert len(waves[0]) == batch_size, (len(waves[0]), batch_size)
if self.normalize_output_wav:
waves = [
(w / abs(w).max(dim=1, keepdim=True)[0] * 0.9).cpu().numpy()
for w in waves
] # list[(batch, sample)]
else:
waves = [w.cpu().numpy() for w in waves]
return waves
@staticmethod
@torch.no_grad()
def normalize_scale(enh_wav, ref_ch_wav):
"""Normalize the energy of enh_wav to match that of ref_ch_wav.
Args:
enh_wav (torch.Tensor): (B, Nsamples)
ref_ch_wav (torch.Tensor): (B, Nsamples)
Returns:
enh_wav (torch.Tensor): (B, Nsamples)
"""
ref_energy = torch.sqrt(torch.mean(ref_ch_wav.pow(2), dim=1))
enh_energy = torch.sqrt(torch.mean(enh_wav.pow(2), dim=1))
return enh_wav * (ref_energy / enh_energy)[:, None]
@torch.no_grad()
def cal_permumation(self, ref_wavs, enh_wavs, criterion="si_snr"):
"""Calculate the permutation between seaprated streams in two adjacent segments.
Args:
ref_wavs (List[torch.Tensor]): [(Batch, Nsamples)]
enh_wavs (List[torch.Tensor]): [(Batch, Nsamples)]
criterion (str): one of ("si_snr", "mse", "corr)
Returns:
perm (torch.Tensor): permutation for enh_wavs (Batch, num_spk)
"""
loss_func = {
"si_snr": self.enh_model.si_snr_loss,
"mse": lambda enh, ref: torch.mean((enh - ref).pow(2), dim=1),
"corr": lambda enh, ref: (
(enh * ref).sum(dim=1)
/ (enh.pow(2).sum(dim=1) * ref.pow(2).sum(dim=1) + EPS)
).clamp(min=EPS, max=1 - EPS),
}[criterion]
_, perm = self.enh_model._permutation_loss(ref_wavs, enh_wavs, loss_func)
return perm
def humanfriendly_or_none(value: str):
if value in ("none", "None", "NONE"):
return None
return humanfriendly.parse_size(value)
def inference(
output_dir: str,
batch_size: int,
dtype: str,
fs: int,
ngpu: int,
seed: int,
num_workers: int,
log_level: Union[int, str],
data_path_and_name_and_type: Sequence[Tuple[str, str, str]],
key_file: Optional[str],
enh_train_config: str,
enh_model_file: str,
allow_variable_data_keys: bool,
segment_size: Optional[float],
hop_size: Optional[float],
normalize_segment_scale: bool,
show_progressbar: bool,
ref_channel: Optional[int],
normalize_output_wav: bool,
):
assert check_argument_types()
if batch_size > 1:
raise NotImplementedError("batch decoding is not implemented")
if ngpu > 1:
raise NotImplementedError("only single GPU decoding is supported")
logging.basicConfig(
level=log_level,
format="%(asctime)s (%(module)s:%(lineno)d) %(levelname)s: %(message)s",
)
if ngpu >= 1:
device = "cuda"
else:
device = "cpu"
# 1. Set random-seed
set_all_random_seed(seed)
# 2. Build separate_speech
separate_speech = SeparateSpeech(
enh_train_config=enh_train_config,
enh_model_file=enh_model_file,
segment_size=segment_size,
hop_size=hop_size,
normalize_segment_scale=normalize_segment_scale,
show_progressbar=show_progressbar,
ref_channel=ref_channel,
normalize_output_wav=normalize_output_wav,
device=device,
dtype=dtype,
)
# 3. Build data-iterator
loader = EnhancementTask.build_streaming_iterator(
data_path_and_name_and_type,
dtype=dtype,
batch_size=batch_size,
key_file=key_file,
num_workers=num_workers,
preprocess_fn=EnhancementTask.build_preprocess_fn(
separate_speech.enh_train_args, False
),
collate_fn=EnhancementTask.build_collate_fn(
separate_speech.enh_train_args, False
),
allow_variable_data_keys=allow_variable_data_keys,
inference=True,
)
# 4. Start for-loop
writers = []
for i in range(separate_speech.num_spk):
writers.append(
SoundScpWriter(f"{output_dir}/wavs/{i + 1}", f"{output_dir}/spk{i + 1}.scp")
)
for keys, batch in loader:
assert isinstance(batch, dict), type(batch)
assert all(isinstance(s, str) for s in keys), keys
_bs = len(next(iter(batch.values())))
assert len(keys) == _bs, f"{len(keys)} != {_bs}"
batch = {k: v for k, v in batch.items() if not k.endswith("_lengths")}
waves = separate_speech(**batch)
for (spk, w) in enumerate(waves):
for b in range(batch_size):
writers[spk][keys[b]] = fs, w[b]
for writer in writers:
writer.close()
def get_parser():
parser = config_argparse.ArgumentParser(
description="Frontend inference",
formatter_class=argparse.ArgumentDefaultsHelpFormatter,
)
# Note(kamo): Use '_' instead of '-' as separator.
# '-' is confusing if written in yaml.
parser.add_argument(
"--log_level",
type=lambda x: x.upper(),
default="INFO",
choices=("CRITICAL", "ERROR", "WARNING", "INFO", "DEBUG", "NOTSET"),
help="The verbose level of logging",
)
parser.add_argument("--output_dir", type=str, required=True)
parser.add_argument(
"--ngpu",
type=int,
default=0,
help="The number of gpus. 0 indicates CPU mode",
)
parser.add_argument("--seed", type=int, default=0, help="Random seed")
parser.add_argument(
"--dtype",
default="float32",
choices=["float16", "float32", "float64"],
help="Data type",
)
parser.add_argument(
"--fs", type=humanfriendly_or_none, default=8000, help="Sampling rate"
)
parser.add_argument(
"--num_workers",
type=int,
default=1,
help="The number of workers used for DataLoader",
)
group = parser.add_argument_group("Input data related")
group.add_argument(
"--data_path_and_name_and_type",
type=str2triple_str,
required=True,
action="append",
)
group.add_argument("--key_file", type=str_or_none)
group.add_argument("--allow_variable_data_keys", type=str2bool, default=False)
group = parser.add_argument_group("Output data related")
group.add_argument(
"--normalize_output_wav",
type=str2bool,
default=False,
help="Whether to normalize the predicted wav to [-1~1]",
)
group = parser.add_argument_group("The model configuration related")
group.add_argument("--enh_train_config", type=str, required=True)
group.add_argument("--enh_model_file", type=str, required=True)
group = parser.add_argument_group("Data loading related")
group.add_argument(
"--batch_size",
type=int,
default=1,
help="The batch size for inference",
)
group = parser.add_argument_group("SeparateSpeech related")
group.add_argument(
"--segment_size",
type=float,
default=None,
help="Segment length in seconds for segment-wise speech enhancement/separation",
)
group.add_argument(
"--hop_size",
type=float,
default=None,
help="Hop length in seconds for segment-wise speech enhancement/separation",
)
group.add_argument(
"--normalize_segment_scale",
type=str2bool,
default=False,
help="Whether to normalize the energy of the separated streams in each segment",
)
group.add_argument(
"--show_progressbar",
type=str2bool,
default=False,
help="Whether to show a progress bar when performing segment-wise speech "
"enhancement/separation",
)
group.add_argument(
"--ref_channel",
type=int,
default=None,
help="If not None, this will overwrite the ref_channel defined in the "
"separator module (for multi-channel speech processing)",
)
return parser
def main(cmd=None):
print(get_commandline_args(), file=sys.stderr)
parser = get_parser()
args = parser.parse_args(cmd)
kwargs = vars(args)
kwargs.pop("config", None)
inference(**kwargs)
if __name__ == "__main__":
main()