# 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 os
from pathlib import Path
import numpy as np
import paddle
import soundfile as sf
import yaml
from paddle import jit
from paddle.static import InputSpec
from timer import timer
from yacs.config import CfgNode
from paddlespeech.t2s.exps.syn_utils import denorm
from paddlespeech.t2s.exps.syn_utils import get_chunks
from paddlespeech.t2s.exps.syn_utils import get_frontend
from paddlespeech.t2s.exps.syn_utils import get_sentences
from paddlespeech.t2s.exps.syn_utils import get_voc_inference
from paddlespeech.t2s.exps.syn_utils import model_alias
from paddlespeech.t2s.exps.syn_utils import run_frontend
from paddlespeech.t2s.exps.syn_utils import voc_to_static
from paddlespeech.t2s.utils import str2bool
from paddlespeech.utils.dynamic_import import dynamic_import
[docs]def evaluate(args):
# 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)
sentences = get_sentences(text_file=args.text, lang=args.lang)
# frontend
frontend = get_frontend(
lang=args.lang,
phones_dict=args.phones_dict,
tones_dict=args.tones_dict)
with open(args.phones_dict, "r") as f:
phn_id = [line.strip().split() for line in f.readlines()]
vocab_size = len(phn_id)
print("vocab_size:", vocab_size)
# acoustic model, only support fastspeech2 here now!
# model: {model_name}_{dataset}
am_name = args.am[:args.am.rindex('_')]
am_dataset = args.am[args.am.rindex('_') + 1:]
odim = am_config.n_mels
am_class = dynamic_import(am_name, model_alias)
am = am_class(idim=vocab_size, odim=odim, **am_config["model"])
am.set_state_dict(paddle.load(args.am_ckpt)["main_params"])
am.eval()
am_mu, am_std = np.load(args.am_stat)
am_mu = paddle.to_tensor(am_mu)
am_std = paddle.to_tensor(am_std)
# am sub layers
am_encoder_infer = am.encoder_infer
am_decoder = am.decoder
am_postnet = am.postnet
# vocoder
voc_inference = get_voc_inference(
voc=args.voc,
voc_config=voc_config,
voc_ckpt=args.voc_ckpt,
voc_stat=args.voc_stat)
# whether dygraph to static
if args.inference_dir:
# fastspeech2 cnndecoder to static
# am.encoder_infer
am_encoder_infer = jit.to_static(
am_encoder_infer, input_spec=[InputSpec([-1], dtype=paddle.int64)])
paddle.jit.save(am_encoder_infer,
os.path.join(args.inference_dir,
args.am + "_am_encoder_infer"))
am_encoder_infer = paddle.jit.load(
os.path.join(args.inference_dir, args.am + "_am_encoder_infer"))
# am.decoder
am_decoder = jit.to_static(
am_decoder,
input_spec=[InputSpec([1, -1, 384], dtype=paddle.float32)])
paddle.jit.save(am_decoder,
os.path.join(args.inference_dir,
args.am + "_am_decoder"))
am_decoder = paddle.jit.load(
os.path.join(args.inference_dir, args.am + "_am_decoder"))
# am.postnet
am_postnet = jit.to_static(
am_postnet,
input_spec=[InputSpec([1, 80, -1], dtype=paddle.float32)])
paddle.jit.save(am_postnet,
os.path.join(args.inference_dir,
args.am + "_am_postnet"))
am_postnet = paddle.jit.load(
os.path.join(args.inference_dir, args.am + "_am_postnet"))
# vocoder
voc_inference = voc_to_static(
voc_inference=voc_inference,
voc=args.voc,
inference_dir=args.inference_dir)
output_dir = Path(args.output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
merge_sentences = True
get_tone_ids = False
N = 0
T = 0
block_size = args.block_size
pad_size = args.pad_size
for utt_id, sentence in sentences:
with timer() as t:
frontend_dict = run_frontend(
frontend=frontend,
text=sentence,
merge_sentences=merge_sentences,
get_tone_ids=get_tone_ids,
lang=args.lang)
phone_ids = frontend_dict['phone_ids']
# merge_sentences=True here, so we only use the first item of phone_ids
phone_ids = phone_ids[0]
with paddle.no_grad():
# acoustic model
orig_hs = am_encoder_infer(phone_ids)
if args.am_streaming:
hss = get_chunks(orig_hs, block_size, pad_size)
chunk_num = len(hss)
mel_list = []
for i, hs in enumerate(hss):
before_outs = am_decoder(hs)
after_outs = before_outs + am_postnet(
before_outs.transpose((0, 2, 1))).transpose(
(0, 2, 1))
normalized_mel = after_outs[0]
sub_mel = denorm(normalized_mel, am_mu, am_std)
# clip output part of pad
if i == 0:
sub_mel = sub_mel[:-pad_size]
elif i == chunk_num - 1:
# 最后一块的右侧一定没有 pad 够
sub_mel = sub_mel[pad_size:]
else:
# 倒数几块的右侧也可能没有 pad 够
sub_mel = sub_mel[pad_size:(block_size + pad_size) -
sub_mel.shape[0]]
mel_list.append(sub_mel)
mel = paddle.concat(mel_list, axis=0)
else:
before_outs = am_decoder(orig_hs)
after_outs = before_outs + am_postnet(
before_outs.transpose((0, 2, 1))).transpose((0, 2, 1))
normalized_mel = after_outs[0]
mel = denorm(normalized_mel, am_mu, am_std)
# 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.shape}, 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) }")
[docs]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=['fastspeech2_csmsc'],
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.")
# vocoder
parser.add_argument(
'--voc',
type=str,
default='pwgan_csmsc',
choices=[
'pwgan_csmsc',
'mb_melgan_csmsc',
'style_melgan_csmsc',
'hifigan_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(
'--lang',
type=str,
default='zh',
help='Choose model language. zh or en')
parser.add_argument(
"--inference_dir",
type=str,
default=None,
help="dir to save inference models")
parser.add_argument(
"--ngpu", type=int, default=1, help="if ngpu == 0, use cpu.")
parser.add_argument(
"--text",
type=str,
help="text to synthesize, a 'utt_id sentence' pair per line.")
# streaming related
parser.add_argument(
"--am_streaming",
type=str2bool,
default=False,
help="whether use streaming acoustic model")
parser.add_argument(
"--block_size", type=int, default=42, help="block size of am streaming")
parser.add_argument(
"--pad_size", type=int, default=12, help="pad size of am streaming")
parser.add_argument("--output_dir", type=str, help="output dir.")
args = parser.parse_args()
return args
[docs]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()