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"""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.pt"
model_weight = p_join(os.path.expanduser('~'), ".cache", "experiment_speaker_embedding", "meta_voice_speaker_encoder.pt.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)
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