#!/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()