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236 lines
7.5 KiB
236 lines
7.5 KiB
# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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import argparse
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from pathlib import Path
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import numpy as np
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import paddle
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import soundfile as sf
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from timer import timer
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from paddlespeech.t2s.exps.syn_utils import get_frontend
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from paddlespeech.t2s.exps.syn_utils import get_sentences
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from paddlespeech.t2s.exps.syn_utils import get_sess
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from paddlespeech.t2s.exps.syn_utils import run_frontend
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from paddlespeech.t2s.utils import str2bool
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def ort_predict(args):
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# frontend
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frontend = get_frontend(
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lang=args.lang,
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phones_dict=args.phones_dict,
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tones_dict=args.tones_dict)
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output_dir = Path(args.output_dir)
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output_dir.mkdir(parents=True, exist_ok=True)
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sentences = get_sentences(text_file=args.text, lang=args.lang)
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am_name = args.am[:args.am.rindex('_')]
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am_dataset = args.am[args.am.rindex('_') + 1:]
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fs = 24000 if am_dataset != 'ljspeech' else 22050
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am_sess = get_sess(
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model_path=str(Path(args.inference_dir) / (args.am + '.onnx')),
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device=args.device,
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cpu_threads=args.cpu_threads,
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use_trt=args.use_trt)
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# vocoder
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voc_sess = get_sess(
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model_path=str(Path(args.inference_dir) / (args.voc + '.onnx')),
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device=args.device,
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cpu_threads=args.cpu_threads,
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use_trt=args.use_trt)
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merge_sentences = True
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# frontend warmup
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# Loading model cost 0.5+ seconds
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if args.lang == 'zh':
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frontend.get_input_ids(
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"你好，欢迎使用飞桨框架进行深度学习研究！", merge_sentences=merge_sentences)
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else:
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frontend.get_input_ids(
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"hello, thank you, thank you very much",
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merge_sentences=merge_sentences)
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# am warmup
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spk_id = [args.spk_id]
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for T in [27, 38, 54]:
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am_input_feed = {}
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if am_name == 'fastspeech2':
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if args.lang == 'en':
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phone_ids = np.random.randint(1, 78, size=(T, ))
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else:
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phone_ids = np.random.randint(1, 266, size=(T, ))
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am_input_feed.update({'text': phone_ids})
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if am_dataset in {"aishell3", "vctk", "mix", "canton"}:
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am_input_feed.update({'spk_id': spk_id})
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elif am_name == 'speedyspeech':
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phone_ids = np.random.randint(1, 92, size=(T, ))
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tone_ids = np.random.randint(1, 5, size=(T, ))
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am_input_feed.update({'phones': phone_ids, 'tones': tone_ids})
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am_sess.run(None, input_feed=am_input_feed)
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# voc warmup
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for T in [227, 308, 544]:
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data = np.random.rand(T, 80).astype("float32")
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voc_sess.run(None, input_feed={"logmel": data})
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print("warm up done!")
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N = 0
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T = 0
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merge_sentences = False
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get_tone_ids = False
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if am_name == 'speedyspeech':
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get_tone_ids = True
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am_input_feed = {}
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for utt_id, sentence in sentences:
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with timer() as t:
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frontend_dict = run_frontend(
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frontend=frontend,
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text=sentence,
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merge_sentences=merge_sentences,
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get_tone_ids=get_tone_ids,
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lang=args.lang)
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phone_ids = frontend_dict['phone_ids']
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flags = 0
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for i in range(len(phone_ids)):
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part_phone_ids = phone_ids[i].numpy()
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if am_name == 'fastspeech2':
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am_input_feed.update({'text': part_phone_ids})
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if am_dataset in {"aishell3", "vctk", "mix", "canton"}:
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am_input_feed.update({'spk_id': spk_id})
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elif am_name == 'speedyspeech':
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part_tone_ids = frontend_dict['tone_ids'][i].numpy()
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am_input_feed.update({
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'phones': part_phone_ids,
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'tones': part_tone_ids
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})
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mel = am_sess.run(output_names=None, input_feed=am_input_feed)
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mel = mel[0]
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wav = voc_sess.run(
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output_names=None, input_feed={'logmel': mel})
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wav = wav[0]
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if flags == 0:
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wav_all = wav
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flags = 1
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else:
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wav_all = np.concatenate([wav_all, wav])
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wav = wav_all
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N += len(wav)
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T += t.elapse
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speed = len(wav) / t.elapse
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rtf = fs / speed
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sf.write(str(output_dir / (utt_id + ".wav")), wav, samplerate=fs)
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print(
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f"{utt_id}, mel: {mel.shape}, wave: {len(wav)}, time: {t.elapse}s, Hz: {speed}, RTF: {rtf}."
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)
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print(f"generation speed: {N / T}Hz, RTF: {fs / (N / T) }")
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def parse_args():
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parser = argparse.ArgumentParser(description="Infernce with onnxruntime.")
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# acoustic model
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parser.add_argument(
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'--am',
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type=str,
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default='fastspeech2_csmsc',
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choices=[
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'fastspeech2_csmsc',
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'fastspeech2_aishell3',
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'fastspeech2_ljspeech',
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'fastspeech2_vctk',
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'speedyspeech_csmsc',
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'fastspeech2_mix',
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'fastspeech2_male-zh',
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'fastspeech2_male-en',
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'fastspeech2_male-mix',
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'fastspeech2_canton',
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],
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help='Choose acoustic model type of tts task.')
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parser.add_argument(
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"--phones_dict", type=str, default=None, help="phone vocabulary file.")
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parser.add_argument(
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"--tones_dict", type=str, default=None, help="tone vocabulary file.")
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parser.add_argument(
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'--spk_id',
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type=int,
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default=0,
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help='spk id for multi speaker acoustic model')
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# voc
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parser.add_argument(
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'--voc',
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type=str,
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default='hifigan_csmsc',
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choices=[
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'pwgan_csmsc',
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'pwgan_aishell3',
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'pwgan_ljspeech',
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'pwgan_vctk',
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'hifigan_csmsc',
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'hifigan_aishell3',
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'hifigan_ljspeech',
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'hifigan_vctk',
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'mb_melgan_csmsc',
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'pwgan_male',
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'hifigan_male',
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],
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help='Choose vocoder type of tts task.')
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# other
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parser.add_argument(
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"--inference_dir", type=str, help="dir to save inference models")
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parser.add_argument(
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"--text",
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type=str,
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help="text to synthesize, a 'utt_id sentence' pair per line")
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parser.add_argument("--output_dir", type=str, help="output dir")
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parser.add_argument(
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'--lang',
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type=str,
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default='zh',
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help='Choose model language. zh or en')
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# inference
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parser.add_argument(
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"--use_trt",
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type=str2bool,
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default=False,
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help="Whether to use inference engin TensorRT.", )
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parser.add_argument(
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"--device",
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default="gpu",
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choices=["gpu", "cpu"],
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help="Device selected for inference.", )
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parser.add_argument('--cpu_threads', type=int, default=1)
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args, _ = parser.parse_known_args()
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return args
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def main():
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args = parse_args()
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paddle.set_device(args.device)
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ort_predict(args)
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if __name__ == "__main__":
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main()
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