allow inference to input and output style vectors

This commit is contained in:
Justin John
2024-01-12 11:31:29 +05:30
committed by GitHub
parent e9a710c75d
commit e2e9fe1589
2 changed files with 436 additions and 426 deletions
+48 -41
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@@ -1,41 +1,48 @@
# FakeYou StyleTTS 2 Inference
import argparse
import numpy as np
import soundfile as sf
from styletts2importable import compute_style, inference
from tortoise.utils.text import split_and_recombine_text
def clsynthesize(text_file_path, voice_path, vcsteps):
# Load the text from the file
with open(text_file_path, 'r', encoding="utf-8") as file:
text = file.read()
# Load the custom voice
custom_voice = compute_style(voice_path)
# Split and recombine long text into manageable chunks
texts = split_and_recombine_text(text)
audios = []
for t in texts:
# Generate audio for each text chunk
audios.append(inference(t, custom_voice, alpha=0.3, beta=0.7, diffusion_steps=vcsteps, embedding_scale=1))
return 24000, np.concatenate(audios)
def main():
parser = argparse.ArgumentParser(description='CLI for StyleTTS Voice Synthesis')
parser.add_argument('--input', type=str, required=True, help='Path to the text file')
parser.add_argument('--voice', type=str, required=True, help='Path to the custom voice audio file')
parser.add_argument('--vcsteps', type=int, required=True, help='Number of diffusion steps')
parser.add_argument('--output', type=str, required=True, help='Output path for the synthesized audio file')
args = parser.parse_args()
sample_rate, audio = clsynthesize(args.input, args.voice, args.vcsteps)
# Save the synthesized audio to the specified output path
sf.write(args.output, audio, sample_rate)
if __name__ == "__main__":
main()
# FakeYou StyleTTS 2 Inference
import os
import torch
import argparse
import numpy as np
import soundfile as sf
from styletts2importable import compute_style, inference
from tortoise.utils.text import split_and_recombine_text
def clsynthesize(text_file_path, voice_path, vcsteps, style_npz_path=None, npz_output_path=None):
# Load the text from the file
with open(text_file_path, 'r', encoding="utf-8") as file:
text = file.read()
device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
if style_npz_path and os.path.exists(style_npz_path):
# Load style vector from NPZ file
style_vector_np = np.load(style_npz_path)['style_vector']
custom_voice = torch.from_numpy(style_vector_np).float().to(device)
elif voice_path:
# Compute and save style vector
custom_voice = compute_style(voice_path, save_npz=True, npz_path=npz_output_path or "default_output_path.npz").to(device)
else:
raise ValueError("Either a voice path or an input style NPZ file must be provided")
texts = split_and_recombine_text(text)
audios = [inference(t, custom_voice, alpha=0.3, beta=0.7, diffusion_steps=vcsteps, embedding_scale=1) for t in texts]
return 24000, np.concatenate(audios)
def main():
parser = argparse.ArgumentParser(description='CLI for StyleTTS Voice Synthesis')
parser.add_argument('--input', type=str, required=True, help='Path to the text file')
parser.add_argument('--voice', type=str, help='Path to the custom voice audio file (optional if using --input-style-npz)')
parser.add_argument('--vcsteps', type=int, required=True, help='Number of diffusion steps')
parser.add_argument('--output', type=str, help='Output path for the synthesized audio file (optional)')
parser.add_argument('--input-style-npz', type=str, help='Path to the input style vector NPZ file')
parser.add_argument('--output-style-npz', type=str, help='Output path for the style vector NPZ file')
args = parser.parse_args()
if args.output:
sample_rate, audio = clsynthesize(args.input, args.voice, args.vcsteps, args.input_style_npz, args.output_style_npz)
sf.write(args.output, audio, sample_rate)
if __name__ == "__main__":
main()
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@@ -1,385 +1,388 @@
import torch
from time import strftime
import os, sys, time
import platform
print("Env vars:")
print(os.environ)
def print_gpu_info():
print('========================================')
print('Python interpreter', sys.executable)
print('PyTorch version', torch.__version__)
print('CUDA Available?', torch.cuda.is_available())
print('CUDA Device count', torch.cuda.device_count())
print('CUDA threads for parallelizing CPU operations', torch.get_num_threads())
print('CUDA architectures library was compiled for', torch.cuda.get_arch_list())
#try:
# from tensorflow.python.client import device_lib
# print('local devices', str(device_lib.list_local_devices()).replace("\n", "\n "))
#except ImportError:
# print('no tensorflow - cannot list devices')
# pass
print('========================================', flush=True)
print_gpu_info(
)
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'
import phonemizer
global_phonemizer = phonemizer.backend.EspeakBackend(language='en-us', preserve_punctuation=True, with_stress=True)
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, use_gruut=False):
text = text.strip()
ps = global_phonemizer.phonemize([text])
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, use_gruut=False):
text = text.strip()
ps = global_phonemizer.phonemize([text])
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, use_gruut=False):
text = text.strip()
ps = global_phonemizer.phonemize([text])
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 = global_phonemizer.phonemize([ref_text])
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
import torch
from time import strftime
import os, sys, time
import platform
print("Env vars:")
print(os.environ)
def print_gpu_info():
print('========================================')
print('Python interpreter', sys.executable)
print('PyTorch version', torch.__version__)
print('CUDA Available?', torch.cuda.is_available())
print('CUDA Device count', torch.cuda.device_count())
print('CUDA threads for parallelizing CPU operations', torch.get_num_threads())
print('CUDA architectures library was compiled for', torch.cuda.get_arch_list())
#try:
# from tensorflow.python.client import device_lib
# print('local devices', str(device_lib.list_local_devices()).replace("\n", "\n "))
#except ImportError:
# print('no tensorflow - cannot list devices')
# pass
print('========================================', flush=True)
print_gpu_info(
)
print("NLTK")
import nltk
nltk.download('punkt')
print("SCIPY")
from scipy.io.wavfile import write
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, save_npz=False, npz_path=None):
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))
style_vector = torch.cat([ref_s, ref_p], dim=1)
# Save style vector to NPZ file if requested
if save_npz and npz_path:
np.savez(npz_path, style_vector=style_vector.cpu().numpy())
return style_vector
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'
import phonemizer
global_phonemizer = phonemizer.backend.EspeakBackend(language='en-us', preserve_punctuation=True, with_stress=True)
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, use_gruut=False):
text = text.strip()
ps = global_phonemizer.phonemize([text])
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, use_gruut=False):
text = text.strip()
ps = global_phonemizer.phonemize([text])
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, use_gruut=False):
text = text.strip()
ps = global_phonemizer.phonemize([text])
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 = global_phonemizer.phonemize([ref_text])
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