mirror of
https://github.com/storytold/storyteller-ml.git
synced 2026-10-11 21:41:18 +00:00
371 lines
13 KiB
Python
371 lines
13 KiB
Python
from cached_path import cached_path
|
|
|
|
from dp.phonemizer import Phonemizer
|
|
print("NLTK")
|
|
import nltk
|
|
nltk.download('punkt')
|
|
print("SCIPY")
|
|
from scipy.io.wavfile import write
|
|
print("TORCH STUFF")
|
|
import torch
|
|
print("START")
|
|
torch.manual_seed(0)
|
|
torch.backends.cudnn.benchmark = False
|
|
torch.backends.cudnn.deterministic = True
|
|
|
|
import random
|
|
random.seed(0)
|
|
|
|
import numpy as np
|
|
np.random.seed(0)
|
|
|
|
# load packages
|
|
import time
|
|
import random
|
|
import yaml
|
|
from munch import Munch
|
|
import numpy as np
|
|
import torch
|
|
from torch import nn
|
|
import torch.nn.functional as F
|
|
import torchaudio
|
|
import librosa
|
|
from nltk.tokenize import word_tokenize
|
|
|
|
from models import *
|
|
from utils import *
|
|
from text_utils import TextCleaner
|
|
textclenaer = TextCleaner()
|
|
|
|
|
|
to_mel = torchaudio.transforms.MelSpectrogram(
|
|
n_mels=80, n_fft=2048, win_length=1200, hop_length=300)
|
|
mean, std = -4, 4
|
|
|
|
def length_to_mask(lengths):
|
|
mask = torch.arange(lengths.max()).unsqueeze(0).expand(lengths.shape[0], -1).type_as(lengths)
|
|
mask = torch.gt(mask+1, lengths.unsqueeze(1))
|
|
return mask
|
|
|
|
def preprocess(wave):
|
|
wave_tensor = torch.from_numpy(wave).float()
|
|
mel_tensor = to_mel(wave_tensor)
|
|
mel_tensor = (torch.log(1e-5 + mel_tensor.unsqueeze(0)) - mean) / std
|
|
return mel_tensor
|
|
|
|
def compute_style(path):
|
|
wave, sr = librosa.load(path, sr=24000)
|
|
audio, index = librosa.effects.trim(wave, top_db=30)
|
|
if sr != 24000:
|
|
audio = librosa.resample(audio, sr, 24000)
|
|
mel_tensor = preprocess(audio).to(device)
|
|
|
|
with torch.no_grad():
|
|
ref_s = model.style_encoder(mel_tensor.unsqueeze(1))
|
|
ref_p = model.predictor_encoder(mel_tensor.unsqueeze(1))
|
|
|
|
return torch.cat([ref_s, ref_p], dim=1)
|
|
|
|
device = 'cpu'
|
|
if torch.cuda.is_available():
|
|
device = 'cuda'
|
|
elif torch.backends.mps.is_available():
|
|
print("MPS would be available but cannot be used rn")
|
|
# device = 'mps'
|
|
|
|
|
|
# global_phonemizer = phonemizer.backend.EspeakBackend(language='en-us', preserve_punctuation=True, with_stress=True)
|
|
phonemizer = Phonemizer.from_checkpoint(str(cached_path('https://public-asai-dl-models.s3.eu-central-1.amazonaws.com/DeepPhonemizer/en_us_cmudict_ipa_forward.pt')))
|
|
|
|
|
|
config = yaml.safe_load(open("Models/LibriTTS/config.yml"))
|
|
|
|
# load pretrained ASR model
|
|
ASR_config = config.get('ASR_config', False)
|
|
ASR_path = config.get('ASR_path', False)
|
|
text_aligner = load_ASR_models(ASR_path, ASR_config)
|
|
|
|
# load pretrained F0 model
|
|
F0_path = config.get('F0_path', False)
|
|
pitch_extractor = load_F0_models(F0_path)
|
|
|
|
# load BERT model
|
|
from Utils.PLBERT.util import load_plbert
|
|
BERT_path = config.get('PLBERT_dir', False)
|
|
plbert = load_plbert(BERT_path)
|
|
|
|
model_params = recursive_munch(config['model_params'])
|
|
model = build_model(model_params, text_aligner, pitch_extractor, plbert)
|
|
_ = [model[key].eval() for key in model]
|
|
_ = [model[key].to(device) for key in model]
|
|
|
|
params_whole = torch.load("Models/LibriTTS/epochs_2nd_00020.pth", map_location='cpu')
|
|
params = params_whole['net']
|
|
|
|
for key in model:
|
|
if key in params:
|
|
print('%s loaded' % key)
|
|
try:
|
|
model[key].load_state_dict(params[key])
|
|
except:
|
|
from collections import OrderedDict
|
|
state_dict = params[key]
|
|
new_state_dict = OrderedDict()
|
|
for k, v in state_dict.items():
|
|
name = k[7:] # remove `module.`
|
|
new_state_dict[name] = v
|
|
# load params
|
|
model[key].load_state_dict(new_state_dict, strict=False)
|
|
# except:
|
|
# _load(params[key], model[key])
|
|
_ = [model[key].eval() for key in model]
|
|
|
|
from Modules.diffusion.sampler import DiffusionSampler, ADPM2Sampler, KarrasSchedule
|
|
|
|
sampler = DiffusionSampler(
|
|
model.diffusion.diffusion,
|
|
sampler=ADPM2Sampler(),
|
|
sigma_schedule=KarrasSchedule(sigma_min=0.0001, sigma_max=3.0, rho=9.0), # empirical parameters
|
|
clamp=False
|
|
)
|
|
|
|
def inference(text, ref_s, alpha = 0.3, beta = 0.7, diffusion_steps=5, embedding_scale=1):
|
|
text = text.strip()
|
|
ps = phonemizer([text], lang='en_us')
|
|
ps = word_tokenize(ps[0])
|
|
ps = ' '.join(ps)
|
|
tokens = textclenaer(ps)
|
|
tokens.insert(0, 0)
|
|
tokens = torch.LongTensor(tokens).to(device).unsqueeze(0)
|
|
|
|
with torch.no_grad():
|
|
input_lengths = torch.LongTensor([tokens.shape[-1]]).to(device)
|
|
text_mask = length_to_mask(input_lengths).to(device)
|
|
|
|
t_en = model.text_encoder(tokens, input_lengths, text_mask)
|
|
bert_dur = model.bert(tokens, attention_mask=(~text_mask).int())
|
|
d_en = model.bert_encoder(bert_dur).transpose(-1, -2)
|
|
|
|
s_pred = sampler(noise = torch.randn((1, 256)).unsqueeze(1).to(device),
|
|
embedding=bert_dur,
|
|
embedding_scale=embedding_scale,
|
|
features=ref_s, # reference from the same speaker as the embedding
|
|
num_steps=diffusion_steps).squeeze(1)
|
|
|
|
|
|
s = s_pred[:, 128:]
|
|
ref = s_pred[:, :128]
|
|
|
|
ref = alpha * ref + (1 - alpha) * ref_s[:, :128]
|
|
s = beta * s + (1 - beta) * ref_s[:, 128:]
|
|
|
|
d = model.predictor.text_encoder(d_en,
|
|
s, input_lengths, text_mask)
|
|
|
|
x, _ = model.predictor.lstm(d)
|
|
duration = model.predictor.duration_proj(x)
|
|
|
|
duration = torch.sigmoid(duration).sum(axis=-1)
|
|
pred_dur = torch.round(duration.squeeze()).clamp(min=1)
|
|
|
|
|
|
pred_aln_trg = torch.zeros(input_lengths, int(pred_dur.sum().data))
|
|
c_frame = 0
|
|
for i in range(pred_aln_trg.size(0)):
|
|
pred_aln_trg[i, c_frame:c_frame + int(pred_dur[i].data)] = 1
|
|
c_frame += int(pred_dur[i].data)
|
|
|
|
# encode prosody
|
|
en = (d.transpose(-1, -2) @ pred_aln_trg.unsqueeze(0).to(device))
|
|
if model_params.decoder.type == "hifigan":
|
|
asr_new = torch.zeros_like(en)
|
|
asr_new[:, :, 0] = en[:, :, 0]
|
|
asr_new[:, :, 1:] = en[:, :, 0:-1]
|
|
en = asr_new
|
|
|
|
F0_pred, N_pred = model.predictor.F0Ntrain(en, s)
|
|
|
|
asr = (t_en @ pred_aln_trg.unsqueeze(0).to(device))
|
|
if model_params.decoder.type == "hifigan":
|
|
asr_new = torch.zeros_like(asr)
|
|
asr_new[:, :, 0] = asr[:, :, 0]
|
|
asr_new[:, :, 1:] = asr[:, :, 0:-1]
|
|
asr = asr_new
|
|
|
|
out = model.decoder(asr,
|
|
F0_pred, N_pred, ref.squeeze().unsqueeze(0))
|
|
|
|
|
|
return out.squeeze().cpu().numpy()[..., :-50] # weird pulse at the end of the model, need to be fixed later
|
|
|
|
def LFinference(text, s_prev, ref_s, alpha = 0.3, beta = 0.7, t = 0.7, diffusion_steps=5, embedding_scale=1):
|
|
text = text.strip()
|
|
ps = phonemizer([text], lang='en_us')
|
|
ps = word_tokenize(ps[0])
|
|
ps = ' '.join(ps)
|
|
ps = ps.replace('``', '"')
|
|
ps = ps.replace("''", '"')
|
|
|
|
tokens = textclenaer(ps)
|
|
tokens.insert(0, 0)
|
|
tokens = torch.LongTensor(tokens).to(device).unsqueeze(0)
|
|
|
|
with torch.no_grad():
|
|
input_lengths = torch.LongTensor([tokens.shape[-1]]).to(device)
|
|
text_mask = length_to_mask(input_lengths).to(device)
|
|
|
|
t_en = model.text_encoder(tokens, input_lengths, text_mask)
|
|
bert_dur = model.bert(tokens, attention_mask=(~text_mask).int())
|
|
d_en = model.bert_encoder(bert_dur).transpose(-1, -2)
|
|
|
|
s_pred = sampler(noise = torch.randn((1, 256)).unsqueeze(1).to(device),
|
|
embedding=bert_dur,
|
|
embedding_scale=embedding_scale,
|
|
features=ref_s, # reference from the same speaker as the embedding
|
|
num_steps=diffusion_steps).squeeze(1)
|
|
|
|
if s_prev is not None:
|
|
# convex combination of previous and current style
|
|
s_pred = t * s_prev + (1 - t) * s_pred
|
|
|
|
s = s_pred[:, 128:]
|
|
ref = s_pred[:, :128]
|
|
|
|
ref = alpha * ref + (1 - alpha) * ref_s[:, :128]
|
|
s = beta * s + (1 - beta) * ref_s[:, 128:]
|
|
|
|
s_pred = torch.cat([ref, s], dim=-1)
|
|
|
|
d = model.predictor.text_encoder(d_en,
|
|
s, input_lengths, text_mask)
|
|
|
|
x, _ = model.predictor.lstm(d)
|
|
duration = model.predictor.duration_proj(x)
|
|
|
|
duration = torch.sigmoid(duration).sum(axis=-1)
|
|
pred_dur = torch.round(duration.squeeze()).clamp(min=1)
|
|
|
|
|
|
pred_aln_trg = torch.zeros(input_lengths, int(pred_dur.sum().data))
|
|
c_frame = 0
|
|
for i in range(pred_aln_trg.size(0)):
|
|
pred_aln_trg[i, c_frame:c_frame + int(pred_dur[i].data)] = 1
|
|
c_frame += int(pred_dur[i].data)
|
|
|
|
# encode prosody
|
|
en = (d.transpose(-1, -2) @ pred_aln_trg.unsqueeze(0).to(device))
|
|
if model_params.decoder.type == "hifigan":
|
|
asr_new = torch.zeros_like(en)
|
|
asr_new[:, :, 0] = en[:, :, 0]
|
|
asr_new[:, :, 1:] = en[:, :, 0:-1]
|
|
en = asr_new
|
|
|
|
F0_pred, N_pred = model.predictor.F0Ntrain(en, s)
|
|
|
|
asr = (t_en @ pred_aln_trg.unsqueeze(0).to(device))
|
|
if model_params.decoder.type == "hifigan":
|
|
asr_new = torch.zeros_like(asr)
|
|
asr_new[:, :, 0] = asr[:, :, 0]
|
|
asr_new[:, :, 1:] = asr[:, :, 0:-1]
|
|
asr = asr_new
|
|
|
|
out = model.decoder(asr,
|
|
F0_pred, N_pred, ref.squeeze().unsqueeze(0))
|
|
|
|
|
|
return out.squeeze().cpu().numpy()[..., :-100], s_pred # weird pulse at the end of the model, need to be fixed later
|
|
|
|
def STinference(text, ref_s, ref_text, alpha = 0.3, beta = 0.7, diffusion_steps=5, embedding_scale=1):
|
|
text = text.strip()
|
|
ps = phonemizer([text], lang='en_us')
|
|
ps = word_tokenize(ps[0])
|
|
ps = ' '.join(ps)
|
|
|
|
tokens = textclenaer(ps)
|
|
tokens.insert(0, 0)
|
|
tokens = torch.LongTensor(tokens).to(device).unsqueeze(0)
|
|
|
|
ref_text = ref_text.strip()
|
|
ps = phonemizer([ref_text], lang='en_us')
|
|
ps = word_tokenize(ps[0])
|
|
ps = ' '.join(ps)
|
|
|
|
ref_tokens = textclenaer(ps)
|
|
ref_tokens.insert(0, 0)
|
|
ref_tokens = torch.LongTensor(ref_tokens).to(device).unsqueeze(0)
|
|
|
|
|
|
with torch.no_grad():
|
|
input_lengths = torch.LongTensor([tokens.shape[-1]]).to(device)
|
|
text_mask = length_to_mask(input_lengths).to(device)
|
|
|
|
t_en = model.text_encoder(tokens, input_lengths, text_mask)
|
|
bert_dur = model.bert(tokens, attention_mask=(~text_mask).int())
|
|
d_en = model.bert_encoder(bert_dur).transpose(-1, -2)
|
|
|
|
ref_input_lengths = torch.LongTensor([ref_tokens.shape[-1]]).to(device)
|
|
ref_text_mask = length_to_mask(ref_input_lengths).to(device)
|
|
ref_bert_dur = model.bert(ref_tokens, attention_mask=(~ref_text_mask).int())
|
|
s_pred = sampler(noise = torch.randn((1, 256)).unsqueeze(1).to(device),
|
|
embedding=bert_dur,
|
|
embedding_scale=embedding_scale,
|
|
features=ref_s, # reference from the same speaker as the embedding
|
|
num_steps=diffusion_steps).squeeze(1)
|
|
|
|
|
|
s = s_pred[:, 128:]
|
|
ref = s_pred[:, :128]
|
|
|
|
ref = alpha * ref + (1 - alpha) * ref_s[:, :128]
|
|
s = beta * s + (1 - beta) * ref_s[:, 128:]
|
|
|
|
d = model.predictor.text_encoder(d_en,
|
|
s, input_lengths, text_mask)
|
|
|
|
x, _ = model.predictor.lstm(d)
|
|
duration = model.predictor.duration_proj(x)
|
|
|
|
duration = torch.sigmoid(duration).sum(axis=-1)
|
|
pred_dur = torch.round(duration.squeeze()).clamp(min=1)
|
|
|
|
|
|
pred_aln_trg = torch.zeros(input_lengths, int(pred_dur.sum().data))
|
|
c_frame = 0
|
|
for i in range(pred_aln_trg.size(0)):
|
|
pred_aln_trg[i, c_frame:c_frame + int(pred_dur[i].data)] = 1
|
|
c_frame += int(pred_dur[i].data)
|
|
|
|
# encode prosody
|
|
en = (d.transpose(-1, -2) @ pred_aln_trg.unsqueeze(0).to(device))
|
|
if model_params.decoder.type == "hifigan":
|
|
asr_new = torch.zeros_like(en)
|
|
asr_new[:, :, 0] = en[:, :, 0]
|
|
asr_new[:, :, 1:] = en[:, :, 0:-1]
|
|
en = asr_new
|
|
|
|
F0_pred, N_pred = model.predictor.F0Ntrain(en, s)
|
|
|
|
asr = (t_en @ pred_aln_trg.unsqueeze(0).to(device))
|
|
if model_params.decoder.type == "hifigan":
|
|
asr_new = torch.zeros_like(asr)
|
|
asr_new[:, :, 0] = asr[:, :, 0]
|
|
asr_new[:, :, 1:] = asr[:, :, 0:-1]
|
|
asr = asr_new
|
|
|
|
out = model.decoder(asr,
|
|
F0_pred, N_pred, ref.squeeze().unsqueeze(0))
|
|
|
|
|
|
return out.squeeze().cpu().numpy()[..., :-50] # weird pulse at the end of the model, need to be fixed later
|
|
print("Time to synthesize!")
|
|
ref_s = compute_style('./voice/voice.wav')
|
|
while True:
|
|
text = input("What to say? > ")
|
|
start = time.time()
|
|
wav = inference(text, ref_s, alpha=0.3, beta=0.7, diffusion_steps=15, embedding_scale=1)
|
|
rtf = (time.time() - start) / (len(wav) / 24000)
|
|
print(f"RTF = {rtf:5f}")
|
|
print(k + ' Synthesized:')
|
|
# display(ipd.Audio(wav, rate=24000, normalize=False))
|
|
write('result.wav', 24000, wav)
|
|
print("Saved to result.wav") |