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synthesize.py
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synthesize.py
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import os
import re
import argparse
from string import punctuation
import torch
import yaml
import numpy as np
from torch.utils.data import DataLoader
from g2p_en import G2p
from pypinyin import pinyin, Style
import audio as Audio
from utils.model import get_model, set_noise_schedule
from utils.tools import to_device, synth_samples, compute_rtf
from dataset import TextDataset
from text import text_to_sequence
from datetime import datetime
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
def read_lexicon(lex_path):
lexicon = {}
with open(lex_path) as f:
for line in f:
temp = re.split(r"\s+", line.strip("\n"))
word = temp[0]
phones = temp[1:]
if word.lower() not in lexicon:
lexicon[word.lower()] = phones
return lexicon
def preprocess_english(text, preprocess_config):
text = text.rstrip(punctuation)
lexicon = read_lexicon(preprocess_config["path"]["lexicon_path"])
g2p = G2p()
phones = []
words = re.split(r"([,;.\-\?\!\s+])", text)
for w in words:
if w.lower() in lexicon:
phones += lexicon[w.lower()]
else:
phones += list(filter(lambda p: p != " ", g2p(w)))
phones = "{" + "}{".join(phones) + "}"
phones = re.sub(r"\{[^\w\s]?\}", "{sp}", phones)
phones = phones.replace("}{", " ")
print("Raw Text Sequence: {}".format(text))
print("Phoneme Sequence: {}".format(phones))
sequence = np.array(
text_to_sequence(
phones, preprocess_config["preprocessing"]["text"]["text_cleaners"]
)
)
return np.array(sequence)
def preprocess_mandarin(text, preprocess_config):
lexicon = read_lexicon(preprocess_config["path"]["lexicon_path"])
phones = []
pinyins = [
p[0]
for p in pinyin(
text, style=Style.TONE3, strict=False, neutral_tone_with_five=True
)
]
for p in pinyins:
if p in lexicon:
phones += lexicon[p]
else:
phones.append("sp")
phones = "{" + " ".join(phones) + "}"
print("Raw Text Sequence: {}".format(text))
print("Phoneme Sequence: {}".format(phones))
sequence = np.array(
text_to_sequence(
phones, preprocess_config["preprocessing"]["text"]["text_cleaners"]
)
)
return np.array(sequence)
def synthesize(model, step, configs, batchs, STFT, control_values):
preprocess_config, model_config, train_config = configs
duration_control = control_values
sampling_rate = preprocess_config["preprocessing"]["audio"]["sampling_rate"]
rtfs = []
for batch in batchs:
batch = to_device(batch, device)
with torch.no_grad():
# Forward
start = datetime.now()
output = model.inference(
*(batch[2:]),
d_control=duration_control,
)
end = datetime.now()
inference_time = (end - start).total_seconds()
rtf = compute_rtf(output, inference_time, sample_rate=sampling_rate)
rtfs.append(rtf)
synth_samples(
output,
batch,
STFT,
preprocess_config,
train_config["path"]["result_path"],
)
print(f'Synthesis Done. RTF estimate: {np.mean(rtfs)} ± {np.std(rtfs)}')
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--restore_step", type=int, required=True)
parser.add_argument(
"--mode",
type=str,
choices=["batch", "single"],
required=True,
help="Synthesize a whole dataset or a single sentence",
)
parser.add_argument(
"--source",
type=str,
default=None,
help="path to a source file with format like train.txt and val.txt, for batch mode only",
)
parser.add_argument(
"--text",
type=str,
default=None,
help="raw text to synthesize, for single-sentence mode only",
)
parser.add_argument(
"--speaker_id",
type=int,
default=0,
help="speaker ID for multi-speaker synthesis, for single-sentence mode only",
)
parser.add_argument(
"-p",
"--preprocess_config",
type=str,
required=True,
help="path to preprocess.yaml",
)
parser.add_argument(
"-m", "--model_config", type=str, required=True, help="path to model.yaml"
)
parser.add_argument(
"-t", "--train_config", type=str, required=True, help="path to train.yaml"
)
parser.add_argument(
"--duration_control",
type=float,
default=1.0,
help="control the speed of the whole utterance, larger value for slower speaking rate",
)
args = parser.parse_args()
# Check source texts
if args.mode == "batch":
assert args.source is not None and args.text is None
if args.mode == "single":
assert args.source is None and args.text is not None
# Read Config
preprocess_config = yaml.load(
open(args.preprocess_config, "r"), Loader=yaml.FullLoader
)
model_config = yaml.load(open(args.model_config, "r"), Loader=yaml.FullLoader)
train_config = yaml.load(open(args.train_config, "r"), Loader=yaml.FullLoader)
configs = (preprocess_config, model_config, train_config)
# Get model
model = get_model(args, configs, device, train=False)
# Set noise schedule
noise_schedule_path = os.path.join(
train_config["path"]["noise_schedule_path"], "{}iters.pt".format(train_config["window"]["noise_iter"])
)
set_noise_schedule(model, noise_schedule_path)
# Build STFT
STFT = Audio.stft.TacotronSTFT(
preprocess_config["preprocessing"]["stft"]["filter_length"],
preprocess_config["preprocessing"]["stft"]["hop_length"],
preprocess_config["preprocessing"]["stft"]["win_length"],
preprocess_config["preprocessing"]["mel"]["n_mel_channels"],
preprocess_config["preprocessing"]["audio"]["sampling_rate"],
preprocess_config["preprocessing"]["mel"]["mel_fmin"],
preprocess_config["preprocessing"]["mel"]["mel_fmax"],
)
# Preprocess texts
if args.mode == "batch":
# Get dataset
dataset = TextDataset(args.source, preprocess_config)
batchs = DataLoader(
dataset,
batch_size=8,
collate_fn=dataset.collate_fn,
)
if args.mode == "single":
ids = raw_texts = [args.text[:100]]
speakers = np.array([args.speaker_id])
if preprocess_config["preprocessing"]["text"]["language"] == "en":
texts = np.array([preprocess_english(args.text, preprocess_config)])
elif preprocess_config["preprocessing"]["text"]["language"] == "zh":
texts = np.array([preprocess_mandarin(args.text, preprocess_config)])
text_lens = np.array([len(texts[0])])
batchs = [(ids, raw_texts, speakers, texts, text_lens, max(text_lens))]
control_values = args.duration_control
synthesize(model, args.restore_step, configs, batchs, STFT, control_values)