"""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/speaker_encoder.pt" model_weight = p_join(os.path.expanduser('~'), ".cache", "experiment_speaker_embedding", "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) # Define the network 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() # Load weight self.load_state_dict(torch.load(model_weight, map_location="cpu")["model_state"], strict=False) # Get the target device 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): # Compute how many frames separate two partial utterances 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)) # Compute the slices 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)) # Evaluate whether extra padding is warranted or not 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") # Wav -> Mel spectrogram 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]) # inference 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)