File size: 16,644 Bytes
ad16788
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
#!/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()