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PaddleSpeech/paddlespeech/t2s/exps/synthesize.py

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# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import argparse
import logging
from pathlib import Path
import jsonlines
import numpy as np
import paddle
import soundfile as sf
import yaml
from timer import timer
from yacs.config import CfgNode
from paddlespeech.t2s.exps.syn_utils import get_am_inference
from paddlespeech.t2s.exps.syn_utils import get_test_dataset
from paddlespeech.t2s.exps.syn_utils import get_voc_inference
from paddlespeech.t2s.utils import str2bool
def evaluate(args):
# dataloader has been too verbose
logging.getLogger("DataLoader").disabled = True
# construct dataset for evaluation
with jsonlines.open(args.test_metadata, 'r') as reader:
test_metadata = list(reader)
# Init body.
with open(args.am_config) as f:
am_config = CfgNode(yaml.safe_load(f))
with open(args.voc_config) as f:
voc_config = CfgNode(yaml.safe_load(f))
print("========Args========")
print(yaml.safe_dump(vars(args)))
print("========Config========")
print(am_config)
print(voc_config)
# acoustic model
am_name = args.am[:args.am.rindex('_')]
am_dataset = args.am[args.am.rindex('_') + 1:]
am_inference = get_am_inference(
am=args.am,
am_config=am_config,
am_ckpt=args.am_ckpt,
am_stat=args.am_stat,
phones_dict=args.phones_dict,
tones_dict=args.tones_dict,
speaker_dict=args.speaker_dict)
test_dataset = get_test_dataset(
test_metadata=test_metadata,
am=args.am,
speaker_dict=args.speaker_dict,
voice_cloning=args.voice_cloning)
# vocoder
voc_inference = get_voc_inference(
voc=args.voc,
voc_config=voc_config,
voc_ckpt=args.voc_ckpt,
voc_stat=args.voc_stat)
output_dir = Path(args.output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
N = 0
T = 0
for datum in test_dataset:
utt_id = datum["utt_id"]
with timer() as t:
with paddle.no_grad():
# acoustic model
if am_name == 'fastspeech2':
phone_ids = paddle.to_tensor(datum["text"])
spk_emb = None
spk_id = None
# multi speaker
if args.voice_cloning and "spk_emb" in datum:
spk_emb = paddle.to_tensor(np.load(datum["spk_emb"]))
elif "spk_id" in datum:
spk_id = paddle.to_tensor(datum["spk_id"])
mel = am_inference(
phone_ids, spk_id=spk_id, spk_emb=spk_emb)
elif am_name == 'speedyspeech':
phone_ids = paddle.to_tensor(datum["phones"])
tone_ids = paddle.to_tensor(datum["tones"])
mel = am_inference(phone_ids, tone_ids)
elif am_name == 'tacotron2':
phone_ids = paddle.to_tensor(datum["text"])
spk_emb = None
# multi speaker
if args.voice_cloning and "spk_emb" in datum:
spk_emb = paddle.to_tensor(np.load(datum["spk_emb"]))
mel = am_inference(phone_ids, spk_emb=spk_emb)
# vocoder
wav = voc_inference(mel)
wav = wav.numpy()
N += wav.size
T += t.elapse
speed = wav.size / t.elapse
rtf = am_config.fs / speed
print(
f"{utt_id}, mel: {mel.shape}, wave: {wav.size}, time: {t.elapse}s, Hz: {speed}, RTF: {rtf}."
)
sf.write(
str(output_dir / (utt_id + ".wav")), wav, samplerate=am_config.fs)
print(f"{utt_id} done!")
print(f"generation speed: {N / T}Hz, RTF: {am_config.fs / (N / T) }")
def parse_args():
# parse args and config
parser = argparse.ArgumentParser(
description="Synthesize with acoustic model & vocoder")
# acoustic model
parser.add_argument(
'--am',
type=str,
default='fastspeech2_csmsc',
choices=[
'speedyspeech_csmsc', 'fastspeech2_csmsc', 'fastspeech2_ljspeech',
'fastspeech2_aishell3', 'fastspeech2_vctk', 'tacotron2_csmsc',
'tacotron2_ljspeech', 'tacotron2_aishell3', 'fastspeech2_mix',
'fastspeech2_canton'
],
help='Choose acoustic model type of tts task.')
parser.add_argument(
'--am_config', type=str, default=None, help='Config of acoustic model.')
parser.add_argument(
'--am_ckpt',
type=str,
default=None,
help='Checkpoint file of acoustic model.')
parser.add_argument(
"--am_stat",
type=str,
default=None,
help="mean and standard deviation used to normalize spectrogram when training acoustic model."
)
parser.add_argument(
"--phones_dict", type=str, default=None, help="phone vocabulary file.")
parser.add_argument(
"--tones_dict", type=str, default=None, help="tone vocabulary file.")
parser.add_argument(
"--speaker_dict", type=str, default=None, help="speaker id map file.")
parser.add_argument(
"--voice-cloning",
type=str2bool,
default=False,
help="whether training voice cloning model.")
# vocoder
parser.add_argument(
'--voc',
type=str,
default='pwgan_csmsc',
choices=[
'pwgan_csmsc', 'pwgan_ljspeech', 'pwgan_aishell3', 'pwgan_vctk',
'mb_melgan_csmsc', 'wavernn_csmsc', 'hifigan_csmsc',
'hifigan_ljspeech', 'hifigan_aishell3', 'hifigan_vctk',
'style_melgan_csmsc'
],
help='Choose vocoder type of tts task.')
parser.add_argument(
'--voc_config', type=str, default=None, help='Config of voc.')
parser.add_argument(
'--voc_ckpt', type=str, default=None, help='Checkpoint file of voc.')
parser.add_argument(
"--voc_stat",
type=str,
default=None,
help="mean and standard deviation used to normalize spectrogram when training voc."
)
# other
parser.add_argument(
"--ngpu", type=int, default=1, help="if ngpu == 0, use cpu.")
parser.add_argument("--test_metadata", type=str, help="test metadata.")
parser.add_argument("--output_dir", type=str, help="output dir.")
args = parser.parse_args()
return args
def main():
args = parse_args()
if args.ngpu == 0:
paddle.set_device("cpu")
elif args.ngpu > 0:
paddle.set_device("gpu")
else:
print("ngpu should >= 0 !")
evaluate(args)
if __name__ == "__main__":
main()