Spaces:
Runtime error
Runtime error
File size: 7,007 Bytes
2cb106d |
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 |
import os
import soundfile as sf
import torch
from torch.optim import SGD
from tqdm import tqdm
from InferenceInterfaces.Meta_FastSpeech2 import Meta_FastSpeech2
from Preprocessing.ArticulatoryCombinedTextFrontend import ArticulatoryCombinedTextFrontend
from Preprocessing.AudioPreprocessor import AudioPreprocessor
from TrainingInterfaces.Text_to_Spectrogram.AutoAligner.Aligner import Aligner
from TrainingInterfaces.Text_to_Spectrogram.FastSpeech2.DurationCalculator import DurationCalculator
from TrainingInterfaces.Text_to_Spectrogram.FastSpeech2.EnergyCalculator import EnergyCalculator
from TrainingInterfaces.Text_to_Spectrogram.FastSpeech2.PitchCalculator import Dio
class UtteranceCloner:
def __init__(self, device):
self.tts = Meta_FastSpeech2(device=device)
self.device = device
torch.hub._validate_not_a_forked_repo = lambda a, b, c: True # torch 1.9 has a bug in the hub loading, this is a workaround
# careful: assumes 16kHz or 8kHz audio
self.silero_model, utils = torch.hub.load(repo_or_dir='snakers4/silero-vad',
model='silero_vad',
force_reload=False,
onnx=False,
verbose=False)
(self.get_speech_timestamps, _, _, _, _) = utils
torch.set_grad_enabled(True) # finding this issue was very infuriating: silero sets
# this to false globally during model loading rather than using inference mode or no_grad
self.silero_model = self.silero_model.to(self.device)
def extract_prosody(self, transcript, ref_audio_path, lang="de", on_line_fine_tune=False):
acoustic_model = Aligner()
acoustic_checkpoint_path = os.path.join("Models", "Aligner", "aligner.pt")
acoustic_model.load_state_dict(torch.load(acoustic_checkpoint_path, map_location='cpu')["asr_model"])
acoustic_model = acoustic_model.to(self.device)
dio = Dio(reduction_factor=1, fs=16000)
energy_calc = EnergyCalculator(reduction_factor=1, fs=16000)
dc = DurationCalculator(reduction_factor=1)
wave, sr = sf.read(ref_audio_path)
tf = ArticulatoryCombinedTextFrontend(language=lang, use_word_boundaries=False)
ap = AudioPreprocessor(input_sr=sr, output_sr=16000, melspec_buckets=80, hop_length=256, n_fft=1024, cut_silence=False)
try:
norm_wave = ap.audio_to_wave_tensor(normalize=True, audio=wave)
except ValueError:
print('Something went wrong, the reference wave might be too short.')
raise RuntimeError
with torch.inference_mode():
speech_timestamps = self.get_speech_timestamps(norm_wave, self.silero_model, sampling_rate=16000)
norm_wave = norm_wave[speech_timestamps[0]['start']:speech_timestamps[-1]['end']]
norm_wave_length = torch.LongTensor([len(norm_wave)])
text = tf.string_to_tensor(transcript, handle_missing=False).squeeze(0)
melspec = ap.audio_to_mel_spec_tensor(audio=norm_wave, normalize=False, explicit_sampling_rate=16000).transpose(0, 1)
melspec_length = torch.LongTensor([len(melspec)]).numpy()
if on_line_fine_tune:
# we fine-tune the aligner for a couple steps using SGD. This makes cloning pretty slow, but the results are greatly improved.
steps = 10
tokens = list() # we need an ID sequence for training rather than a sequence of phonological features
for vector in text:
for phone in tf.phone_to_vector:
if vector.numpy().tolist() == tf.phone_to_vector[phone]:
tokens.append(tf.phone_to_id[phone])
tokens = torch.LongTensor(tokens)
tokens = tokens.squeeze().to(self.device)
tokens_len = torch.LongTensor([len(tokens)]).to(self.device)
mel = melspec.unsqueeze(0).to(self.device)
mel.requires_grad = True
mel_len = torch.LongTensor([len(mel[0])]).to(self.device)
# actual fine-tuning starts here
optim_asr = SGD(acoustic_model.parameters(), lr=0.1)
acoustic_model.train()
for _ in tqdm(list(range(steps))):
pred = acoustic_model(mel)
loss = acoustic_model.ctc_loss(pred.transpose(0, 1).log_softmax(2), tokens, mel_len, tokens_len)
optim_asr.zero_grad()
loss.backward()
torch.nn.utils.clip_grad_norm_(acoustic_model.parameters(), 1.0)
optim_asr.step()
acoustic_model.eval()
torch.save({"asr_model": acoustic_model.state_dict()},
os.path.join(os.path.join("Models", "Aligner", "aligner.pt")))
alignment_path = acoustic_model.inference(mel=melspec.to(self.device),
tokens=text.to(self.device),
return_ctc=False)
duration = dc(torch.LongTensor(alignment_path), vis=None).cpu()
energy = energy_calc(input_waves=norm_wave.unsqueeze(0),
input_waves_lengths=norm_wave_length,
feats_lengths=melspec_length,
durations=duration.unsqueeze(0),
durations_lengths=torch.LongTensor([len(duration)]))[0].squeeze(0).cpu()
pitch = dio(input_waves=norm_wave.unsqueeze(0),
input_waves_lengths=norm_wave_length,
feats_lengths=melspec_length,
durations=duration.unsqueeze(0),
durations_lengths=torch.LongTensor([len(duration)]))[0].squeeze(0).cpu()
return duration, pitch, energy, speech_timestamps[0]['start'], speech_timestamps[-1]['end']
def clone_utterance(self,
path_to_reference_audio,
reference_transcription,
clone_speaker_identity=True,
lang="en"):
if clone_speaker_identity:
self.tts.set_utterance_embedding(path_to_reference_audio=path_to_reference_audio)
duration, pitch, energy, silence_frames_start, silence_frames_end = self.extract_prosody(reference_transcription,
path_to_reference_audio,
lang=lang)
self.tts.set_language(lang)
start_sil = torch.zeros([silence_frames_start]).to(self.device)
end_sil = torch.zeros([silence_frames_end]).to(self.device)
cloned_speech = self.tts(reference_transcription, view=False, durations=duration, pitch=pitch, energy=energy)
cloned_utt = torch.cat((start_sil, cloned_speech, end_sil), dim=0)
return cloned_utt.cpu()
|