| """Speaker embedding obtained via speaker verification training. |
| - feature dimension: 256 |
| - source: https://github.com/metavoiceio/metavoice-src |
| """ |
| import os |
| import subprocess |
| from os.path import join as p_join |
| from typing import Optional |
|
|
| import librosa |
| from librosa import feature |
| import numpy as np |
| import torch |
| from torch import nn |
|
|
|
|
| checkpoint_url = "https://huggingface.co/datasets/asahi417/experiment-speaker-embedding/resolve/main/meta_voice_speaker_encoder.pt" |
| model_weight = p_join(os.path.expanduser('~'), ".cache", "experiment_speaker_embedding", "meta_voice_speaker_encoder.pt") |
|
|
|
|
| def wget(url: str, output_file: Optional[str] = None): |
| os.makedirs(os.path.dirname(output_file), exist_ok=True) |
| subprocess.run(["wget", url, "-O", output_file]) |
| if not os.path.exists(output_file): |
| raise ValueError(f"failed to download {url}") |
|
|
|
|
| class MetaVoiceSE(nn.Module): |
|
|
| mel_window_length = 25 |
| mel_window_step = 10 |
| mel_n_channels = 40 |
| sampling_rate = 16000 |
| partials_n_frames = 160 |
| model_hidden_size = 256 |
| model_embedding_size = 256 |
| model_num_layers = 3 |
|
|
| def __init__(self): |
| super().__init__() |
| if not os.path.exists(model_weight): |
| wget(checkpoint_url, model_weight) |
| |
| self.lstm = nn.LSTM(self.mel_n_channels, self.model_hidden_size, self.model_num_layers, batch_first=True) |
| self.linear = nn.Linear(self.model_hidden_size, self.model_embedding_size) |
| self.relu = nn.ReLU() |
| |
| self.load_state_dict(torch.load(model_weight, map_location="cpu")["model_state"], strict=False) |
| |
| self.device = torch.device("cuda" if torch.cuda.is_available() else "cpu") |
| self.to(self.device) |
| self.eval() |
|
|
| def compute_partial_slices(self, n_samples: int, rate, min_coverage): |
| |
| samples_per_frame = int((self.sampling_rate * self.mel_window_step / 1000)) |
| n_frames = int(np.ceil((n_samples + 1) / samples_per_frame)) |
| frame_step = int(np.round((self.sampling_rate / rate) / samples_per_frame)) |
| |
| wav_slices, mel_slices = [], [] |
| steps = max(1, n_frames - self.partials_n_frames + frame_step + 1) |
| for i in range(0, steps, frame_step): |
| mel_range = np.array([i, i + self.partials_n_frames]) |
| wav_range = mel_range * samples_per_frame |
| mel_slices.append(slice(*mel_range)) |
| wav_slices.append(slice(*wav_range)) |
| |
| last_wav_range = wav_slices[-1] |
| coverage = (n_samples - last_wav_range.start) / (last_wav_range.stop - last_wav_range.start) |
| if coverage < min_coverage and len(mel_slices) > 1: |
| return wav_slices[:-1], mel_slices[:-1] |
| return wav_slices, mel_slices |
|
|
| def get_speaker_embedding(self, |
| wav: np.ndarray, |
| sampling_rate: Optional[int] = None, |
| rate: float = 1.3, |
| min_coverage: float = 0.75) -> np.ndarray: |
| if sampling_rate != self.sampling_rate: |
| wav = librosa.resample(wav, orig_sr=sampling_rate, target_sr=self.sampling_rate) |
| wav, _ = librosa.effects.trim(wav, top_db=20) |
| wav_slices, mel_slices = self.compute_partial_slices(len(wav), rate, min_coverage) |
| max_wave_length = wav_slices[-1].stop |
| if max_wave_length >= len(wav): |
| wav = np.pad(wav, (0, max_wave_length - len(wav)), "constant") |
| |
| frames = feature.melspectrogram( |
| y=wav, |
| sr=self.sampling_rate, |
| n_fft=int(self.sampling_rate * self.mel_window_length / 1000), |
| hop_length=int(self.sampling_rate * self.mel_window_step / 1000), |
| n_mels=self.mel_n_channels, |
| ) |
| mel = frames.astype(np.float32).T |
| mel = np.array([mel[s] for s in mel_slices]) |
| |
| with torch.no_grad(): |
| mel = torch.from_numpy(mel).to(self.device) |
| _, (hidden, _) = self.lstm(mel) |
| embeds_raw = self.relu(self.linear(hidden[-1])) |
| partial_embeds = embeds_raw / torch.norm(embeds_raw, dim=1, keepdim=True) |
| partial_embeds = partial_embeds.cpu().numpy() |
| raw_embed = np.mean(partial_embeds, axis=0) |
| return raw_embed / np.linalg.norm(raw_embed, 2) |
|
|