coqui-tts/TTS/bin/synthesize.py

357 lines
12 KiB
Python
Executable File

#!/usr/bin/env python3
# -*- coding: utf-8 -*-
import argparse
import json
import os
import sys
import string
import time
from argparse import RawTextHelpFormatter
# pylint: disable=redefined-outer-name, unused-argument
from pathlib import Path
import numpy as np
import torch
from TTS.tts.utils.generic_utils import is_tacotron, setup_model
from TTS.tts.utils.synthesis import synthesis
from TTS.tts.utils.text.symbols import make_symbols, phonemes, symbols
from TTS.tts.utils.io import load_checkpoint
from TTS.utils.audio import AudioProcessor
from TTS.utils.io import load_config
from TTS.utils.manage import ModelManager
from TTS.vocoder.utils.generic_utils import setup_generator, interpolate_vocoder_input
def str2bool(v):
if isinstance(v, bool):
return v
if v.lower() in ('yes', 'true', 't', 'y', '1'):
return True
elif v.lower() in ('no', 'false', 'f', 'n', '0'):
return False
else:
raise argparse.ArgumentTypeError('Boolean value expected.')
def load_tts_model(model_path, config_path, use_cuda, speakers_json=None, speaker_idx=None):
global phonemes
global symbols
# load the config
model_config = load_config(config_path)
# load the audio processor
ap = AudioProcessor(**model_config.audio)
# if the vocabulary was passed, replace the default
if 'characters' in model_config.keys():
symbols, phonemes = make_symbols(**model_config.characters)
# load speakers
speaker_embedding = None
speaker_embedding_dim = None
num_speakers = 0
if speakers_json is not None:
speaker_mapping = json.load(open(speakers_json, 'r'))
num_speakers = len(speaker_mapping)
if model_config.use_external_speaker_embedding_file:
if speaker_idx is not None:
speaker_embedding = speaker_mapping[speaker_idx]['embedding']
else: # if speaker_idx is not specificated use the first sample in speakers.json
speaker_embedding = speaker_mapping[list(speaker_mapping.keys())[0]]['embedding']
speaker_embedding_dim = len(speaker_embedding)
# load tts model
num_chars = len(phonemes) if model_config.use_phonemes else len(symbols)
model = setup_model(num_chars, num_speakers, model_config, speaker_embedding_dim)
model.load_checkpoint(model_config, model_path, eval=True)
if use_cuda:
model.cuda()
return model, model_config, ap, speaker_embedding
def load_vocoder_model(model_path, config_path, use_cuda):
vocoder_config = load_config(vocoder_config_path)
vocoder_ap = AudioProcessor(**vocoder_config['audio'])
vocoder_model = setup_generator(vocoder_config)
vocoder_model.load_checkpoint(vocoder_config, model_path, eval=True)
if use_cuda:
vocoder_model.cuda()
return vocoder_model, vocoder_config, vocoder_ap
def tts(model,
vocoder_model,
text,
model_config,
vocoder_config,
use_cuda,
ap,
vocoder_ap,
use_gl,
speaker_fileid,
speaker_embedding=None,
gst_style=None):
t_1 = time.time()
waveform, _, _, mel_postnet_spec, _, _ = synthesis(
model,
text,
model_config,
use_cuda,
ap,
speaker_fileid,
gst_style,
False,
model_config.enable_eos_bos_chars,
use_gl,
speaker_embedding=speaker_embedding)
# grab spectrogram (thx to the nice guys at mozilla discourse for codesnippet)
if args.save_spectogram:
spec_file_name = args.text.replace(" ", "_")[0:10]
spec_file_name = spec_file_name.translate(
str.maketrans('', '', string.punctuation.replace('_', ''))) + '.npy'
spec_file_name = os.path.join(args.out_path, spec_file_name)
spectrogram = mel_postnet_spec.T
spectrogram = spectrogram[0]
np.save(spec_file_name, spectrogram)
print(" > Saving raw spectogram to " + spec_file_name)
# convert linear spectrogram to melspectrogram for tacotron
if model_config.model == "Tacotron" and not use_gl:
mel_postnet_spec = ap.out_linear_to_mel(mel_postnet_spec.T)
# run vocoder_model
if not use_gl:
# denormalize tts output based on tts audio config
mel_postnet_spec = ap._denormalize(mel_postnet_spec.T).T
device_type = "cuda" if use_cuda else "cpu"
# renormalize spectrogram based on vocoder config
vocoder_input = vocoder_ap._normalize(mel_postnet_spec.T)
# compute scale factor for possible sample rate mismatch
scale_factor = [1, vocoder_config['audio']['sample_rate'] / ap.sample_rate]
if scale_factor[1] != 1:
print(" > interpolating tts model output.")
vocoder_input = interpolate_vocoder_input(scale_factor, vocoder_input)
else:
vocoder_input = torch.tensor(vocoder_input).unsqueeze(0)
# run vocoder model
# [1, T, C]
waveform = vocoder_model.inference(vocoder_input.to(device_type))
if use_cuda and not use_gl:
waveform = waveform.cpu()
if not use_gl:
waveform = waveform.numpy()
waveform = waveform.squeeze()
rtf = (time.time() - t_1) / (len(waveform) / ap.sample_rate)
tps = (time.time() - t_1) / len(waveform)
print(" > Run-time: {}".format(time.time() - t_1))
print(" > Real-time factor: {}".format(rtf))
print(" > Time per step: {}".format(tps))
return waveform
if __name__ == "__main__":
parser = argparse.ArgumentParser(description='''Synthesize speech on command line.\n\n'''
'''You can either use your trained model or choose a model from the provided list.\n'''
'''
Example runs:
# list provided models
./TTS/bin/synthesize.py --list_models
# run a model from the list
./TTS/bin/synthesize.py --text "Text for TTS" --model_name "<language>/<dataset>/<model_name>" --vocoder_name "<language>/<dataset>/<model_name>" --output_path
# run your own TTS model (Using Griffin-Lim Vocoder)
./TTS/bin/synthesize.py --text "Text for TTS" --model_path path/to/model.pth.tar --config_path path/to/config.json --out_path output/path/speech.wav
# run your own TTS and Vocoder models
./TTS/bin/synthesize.py --text "Text for TTS" --model_path path/to/config.json --config_path path/to/model.pth.tar --out_path output/path/speech.wav
--vocoder_path path/to/vocoder.pth.tar --vocoder_config_path path/to/vocoder_config.json
''',
formatter_class=RawTextHelpFormatter)
parser.add_argument(
'--list_models',
type=str2bool,
nargs='?',
const=True,
default=False,
help='list available pre-trained tts and vocoder models.'
)
parser.add_argument(
'--text',
type=str,
default=None,
help='Text to generate speech.'
)
# Args for running pre-trained TTS models.
parser.add_argument(
'--model_name',
type=str,
default=None,
help=
'Name of one of the pre-trained tts models in format <language>/<dataset>/<model_name>'
)
parser.add_argument(
'--vocoder_name',
type=str,
default=None,
help=
'Name of one of the pre-trained vocoder models in format <language>/<dataset>/<model_name>'
)
# Args for running custom models
parser.add_argument(
'--config_path',
default=None,
type=str,
help='Path to model config file.'
)
parser.add_argument(
'--model_path',
type=str,
default=None,
help='Path to model file.',
)
parser.add_argument(
'--out_path',
type=str,
default=Path(__file__).resolve().parent,
help='Path to save final wav file. Wav file will be named as the given text.',
)
parser.add_argument(
'--use_cuda',
type=bool,
help='Run model on CUDA.',
default=False
)
parser.add_argument(
'--vocoder_path',
type=str,
help=
'Path to vocoder model file. If it is not defined, model uses GL as vocoder. Please make sure that you installed vocoder library before (WaveRNN).',
default=None,
)
parser.add_argument(
'--vocoder_config_path',
type=str,
help='Path to vocoder model config file.',
default=None)
# args for multi-speaker synthesis
parser.add_argument(
'--speakers_json',
type=str,
help="JSON file for multi-speaker model.",
default=None)
parser.add_argument(
'--speaker_idx',
type=str,
help="if the tts model is trained with x-vectors, then speaker_idx is a file present in speakers.json else speaker_idx is the speaker id corresponding to a speaker in the speaker embedding layer.",
default=None)
parser.add_argument(
'--gst_style',
help="Wav path file for GST stylereference.",
default=None)
# aux args
parser.add_argument(
'--save_spectogram',
type=bool,
help="If true save raw spectogram for further (vocoder) processing in out_path.",
default=False)
args = parser.parse_args()
# load model manager
path = Path(__file__).parent / "../../.models.json"
manager = ModelManager(path)
model_path = None
vocoder_path = None
model = None
vocoder_model = None
vocoder_config = None
vocoder_ap = None
# CASE1: list pre-trained TTS models
if args.list_models:
manager.list_models()
sys.exit()
# CASE2: load pre-trained models
if args.model_name is not None:
model_path, config_path = manager.download_model(args.model_name)
if args.vocoder_name is not None:
vocoder_path, vocoder_config_path = manager.download_model(args.vocoder_name)
# CASE3: load custome models
if args.model_path is not None:
model_path = args.model_path
config_path = args.config_path
if args.vocoder_path is not None:
vocoder_path = args.vocoder_path
vocoder_config_path = args.vocoder_config_path
# RUN THE SYNTHESIS
# load models
model, model_config, ap, speaker_embedding = load_tts_model(model_path, config_path, args.use_cuda, args.speaker_idx)
if vocoder_path is not None:
vocoder_model, vocoder_config, vocoder_ap = load_vocoder_model(vocoder_path, vocoder_config_path, use_cuda=args.use_cuda)
use_griffin_lim = vocoder_path is None
print(" > Text: {}".format(args.text))
# handle multi-speaker setting
if not model_config.use_external_speaker_embedding_file and args.speaker_idx is not None:
if args.speaker_idx.isdigit():
args.speaker_idx = int(args.speaker_idx)
else:
args.speaker_idx = None
else:
args.speaker_idx = None
if args.gst_style is None:
if 'gst' in model_config.keys() and model_config.gst['gst_style_input'] is not None:
gst_style = model_config.gst['gst_style_input']
else:
gst_style = None
else:
# check if gst_style string is a dict, if is dict convert else use string
try:
gst_style = json.loads(args.gst_style)
if max(map(int, gst_style.keys())) >= model_config.gst['gst_style_tokens']:
raise RuntimeError("The highest value of the gst_style dictionary key must be less than the number of GST Tokens, \n Highest dictionary key value: {} \n Number of GST tokens: {}".format(max(map(int, gst_style.keys())), model_config.gst['gst_style_tokens']))
except ValueError:
gst_style = args.gst_style
# kick it
wav = tts(model,
vocoder_model,
args.text,
model_config,
vocoder_config,
args.use_cuda,
ap,
vocoder_ap,
use_griffin_lim,
args.speaker_idx,
speaker_embedding=speaker_embedding,
gst_style=gst_style)
# save the results
file_name = args.text.replace(" ", "_")[0:20]
file_name = file_name.translate(
str.maketrans('', '', string.punctuation.replace('_', ''))) + '.wav'
out_path = os.path.join(args.out_path, file_name)
print(" > Saving output to {}".format(out_path))
ap.save_wav(wav, out_path)