Source code for paddlespeech.t2s.exps.synthesize_e2e

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# Licensed under the Apache License, Version 2.0 (the "License");
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#     http://www.apache.org/licenses/LICENSE-2.0
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import argparse
from pathlib import Path
from pprint import pprint

import paddle
import soundfile as sf
import yaml
from timer import timer
from yacs.config import CfgNode

from paddlespeech.t2s.exps.syn_utils import am_to_static
from paddlespeech.t2s.exps.syn_utils import get_am_inference
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_sentences_svs
from paddlespeech.t2s.exps.syn_utils import get_voc_inference
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


[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) # frontend frontend = get_frontend( lang=args.lang, phones_dict=args.phones_dict, tones_dict=args.tones_dict, pinyin_phone=args.pinyin_phone, use_rhy=args.use_rhy) print("frontend done!") # 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, speech_stretchs=args.speech_stretchs, ) print("acoustic model done!") # vocoder voc_inference = get_voc_inference( voc=args.voc, voc_config=voc_config, voc_ckpt=args.voc_ckpt, voc_stat=args.voc_stat) print("voc done!") # whether dygraph to static if args.inference_dir: print("convert am and voc to static model.") # acoustic model am_inference = am_to_static( am_inference=am_inference, am=args.am, inference_dir=args.inference_dir, speaker_dict=args.speaker_dict) # 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 = False # Avoid not stopping at the end of a sub sentence when tacotron2_ljspeech dygraph to static graph # but still not stopping in the end (NOTE by yuantian01 Feb 9 2022) if am_name == 'tacotron2': merge_sentences = True get_tone_ids = False if am_name == 'speedyspeech': get_tone_ids = True # wav samples N = 0 # inference time cost T = 0 # [(uid, text), ] if am_name == 'diffsinger': sentences = get_sentences_svs(text_file=args.text) else: sentences = get_sentences(text_file=args.text, lang=args.lang) for utt_id, sentence in sentences: print(f"{utt_id} {sentence}") with timer() as t: if am_name == "diffsinger": text = "" svs_input = sentence else: text = sentence svs_input = None # frontend frontend_dict = run_frontend( frontend=frontend, text=text, merge_sentences=merge_sentences, get_tone_ids=get_tone_ids, lang=args.lang, svs_input=svs_input) phone_ids = frontend_dict['phone_ids'] # pprint(f"{utt_id} {phone_ids}") with paddle.no_grad(): flags = 0 for i in range(len(phone_ids)): # sub phone, split by `sp` or punctuation. part_phone_ids = phone_ids[i] # acoustic model if am_name == 'fastspeech2': # multi speaker if am_dataset in {"aishell3", "vctk", "mix", "canton"}: # multi-speaker spk_id = paddle.to_tensor([args.spk_id]) mel = am_inference(part_phone_ids, spk_id) else: # single-speaker mel = am_inference(part_phone_ids) elif am_name == 'speedyspeech': part_tone_ids = frontend_dict['tone_ids'][i] if am_dataset in {"aishell3", "vctk", "mix"}: # multi-speaker spk_id = paddle.to_tensor([args.spk_id]) mel = am_inference(part_phone_ids, part_tone_ids, spk_id) else: # single-speaker mel = am_inference(part_phone_ids, part_tone_ids) elif am_name == 'tacotron2': mel = am_inference(part_phone_ids) elif am_name == 'diffsinger': part_note_ids = frontend_dict['note_ids'][i] part_note_durs = frontend_dict['note_durs'][i] part_is_slurs = frontend_dict['is_slurs'][i] mel = am_inference( text=part_phone_ids, note=part_note_ids, note_dur=part_note_durs, is_slur=part_is_slurs, ) # vocoder wav = voc_inference(mel) if flags == 0: wav_all = wav flags = 1 else: wav_all = paddle.concat([wav_all, wav]) wav = wav_all.numpy() N += wav.size T += t.elapse # samples per second speed = wav.size / t.elapse # generate one second wav need `RTF` seconds 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=[ 'speedyspeech_csmsc', 'speedyspeech_aishell3', 'fastspeech2_csmsc', 'fastspeech2_ljspeech', 'fastspeech2_aishell3', 'fastspeech2_vctk', 'tacotron2_csmsc', 'tacotron2_ljspeech', 'fastspeech2_mix', 'fastspeech2_canton', 'fastspeech2_male-zh', 'fastspeech2_male-en', 'fastspeech2_male-mix', 'diffsinger_opencpop', ], 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( '--spk_id', type=int, default=0, help='spk id for multi speaker acoustic model') # vocoder parser.add_argument( '--voc', type=str, default='pwgan_csmsc', choices=[ 'pwgan_csmsc', 'pwgan_ljspeech', 'pwgan_aishell3', 'pwgan_vctk', 'mb_melgan_csmsc', 'style_melgan_csmsc', 'hifigan_csmsc', 'hifigan_ljspeech', 'hifigan_aishell3', 'hifigan_vctk', 'wavernn_csmsc', 'pwgan_male', 'hifigan_male', 'pwgan_opencpop', 'hifigan_opencpop', ], 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', choices=['zh', 'en', 'mix', 'canton', 'sing'], help='Choose model language. zh or en or mix') 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.") parser.add_argument("--output_dir", type=str, help="output dir.") parser.add_argument( "--use_rhy", type=str2bool, default=False, help="run rhythm frontend or not") parser.add_argument( "--pinyin_phone", type=str, default=None, help="pinyin to phone map file, using on sing_frontend.") parser.add_argument( "--speech_stretchs", type=str, default=None, help="The min and max values of the mel spectrum, using on diffusion of diffsinger." ) 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()