# Copyright (c) 2022 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 from pathlib import Path import soundfile as sf from timer import timer from paddlespeech.t2s.exps.lite_syn_utils import get_lite_am_output from paddlespeech.t2s.exps.lite_syn_utils import get_lite_predictor from paddlespeech.t2s.exps.syn_utils import get_frontend from paddlespeech.t2s.exps.syn_utils import get_sentences def parse_args(): parser = argparse.ArgumentParser( description="Paddle Infernce with acoustic model & vocoder.") # acoustic model parser.add_argument( '--am', type=str, default='vits_csmsc', choices=[ 'vits_csmsc', 'vits_aishell3', ], help='Choose acoustic model type of tts task.') parser.add_argument( "--phones_dict", type=str, default=None, help="phone vocabulary file.") parser.add_argument( "--speaker_dict", type=str, default=None, help="speaker id map file.") parser.add_argument( '--spk_id', type=int, default=0, help='spk id for multi speaker acoustic model') # other parser.add_argument( '--lang', type=str, default='zh', help='Choose model language. zh or en or mix') parser.add_argument( "--text", type=str, help="text to synthesize, a 'utt_id sentence' pair per line") parser.add_argument( "--add-blank", type=str2bool, default=True, help="whether to add blank between phones") parser.add_argument( "--inference_dir", type=str, help="dir to save inference models") parser.add_argument("--output_dir", type=str, help="output dir") args, _ = parser.parse_known_args() return args # only inference for models trained with csmsc now def main(): args = parse_args() # frontend frontend = get_frontend( lang=args.lang, phones_dict=args.phones_dict, tones_dict=args.tones_dict) # am_predictor am_predictor = get_lite_predictor( model_dir=args.inference_dir, model_file=args.am + "_x86.nb") # model: {model_name}_{dataset} am_dataset = args.am[args.am.rindex('_') + 1:] output_dir = Path(args.output_dir) output_dir.mkdir(parents=True, exist_ok=True) sentences = get_sentences(text_file=args.text, lang=args.lang) merge_sentences = True add_blank = args.add_blank fs = 22050 # warmup for utt_id, sentence in sentences[:3]: with timer() as t: wav = get_lite_am_output( input=sentence, am_predictor=am_predictor, am=args.am, frontend=frontend, lang=args.lang, merge_sentences=merge_sentences, speaker_dict=args.speaker_dict, spk_id=args.spk_id, add_blank=add_blank) speed = wav.size / t.elapse rtf = fs / speed print( f"{utt_id}, wave: {wav.shape}, time: {t.elapse}s, Hz: {speed}, RTF: {rtf}." ) print("warm up done!") N = 0 T = 0 for utt_id, sentence in sentences: with timer() as t: wav = get_lite_am_output( input=sentence, am_predictor=am_predictor, am=args.am, frontend=frontend, lang=args.lang, merge_sentences=merge_sentences, speaker_dict=args.speaker_dict, spk_id=args.spk_id, add_blank=add_blank) N += wav.size T += t.elapse speed = wav.size / t.elapse rtf = fs / speed sf.write(output_dir / (utt_id + ".wav"), wav, samplerate=fs) print( f"{utt_id}, wave: {wav.shape}, time: {t.elapse}s, Hz: {speed}, RTF: {rtf}." ) print(f"{utt_id} done!") print(f"generation speed: {N / T}Hz, RTF: {fs / (N / T) }") if __name__ == "__main__": main()