moe-tts / app.py
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import json
import os
import re
import librosa
import numpy as np
import torch
from torch import no_grad, LongTensor
import commons
import utils
import gradio as gr
from models import SynthesizerTrn
from text import text_to_sequence, _clean_text
from mel_processing import spectrogram_torch
limitation = os.getenv("SYSTEM") == "spaces" # limit text and audio length in huggingface spaces
def get_text(text, hps, is_phoneme):
text_norm = text_to_sequence(text, hps.symbols, [] if is_phoneme else hps.data.text_cleaners)
if hps.data.add_blank:
text_norm = commons.intersperse(text_norm, 0)
text_norm = LongTensor(text_norm)
return text_norm
def create_tts_fn(model, hps, speaker_ids):
def tts_fn(text, speaker, speed, is_phoneme):
if limitation:
text_len = len(text)
max_len = 60
if is_phoneme:
max_len *= 3
else:
if len(hps.data.text_cleaners) > 0 and hps.data.text_cleaners[0] == "zh_ja_mixture_cleaners":
text_len = len(re.sub("(\[ZH\]|\[JA\])", "", text))
if text_len > max_len:
return "Error: Text is too long", None
speaker_id = speaker_ids[speaker]
stn_tst = get_text(text, hps, is_phoneme)
with no_grad():
x_tst = stn_tst.unsqueeze(0)
x_tst_lengths = LongTensor([stn_tst.size(0)])
sid = LongTensor([speaker_id])
audio = model.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=.667, noise_scale_w=0.8,
length_scale=1.0 / speed)[0][0, 0].data.cpu().float().numpy()
del stn_tst, x_tst, x_tst_lengths, sid
return "Success", (hps.data.sampling_rate, audio)
return tts_fn
def create_vc_fn(model, hps, speaker_ids):
def vc_fn(original_speaker, target_speaker, input_audio):
if input_audio is None:
return "You need to upload an audio", None
sampling_rate, audio = input_audio
duration = audio.shape[0] / sampling_rate
if limitation and duration > 15:
return "Error: Audio is too long", None
original_speaker_id = speaker_ids[original_speaker]
target_speaker_id = speaker_ids[target_speaker]
audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
if len(audio.shape) > 1:
audio = librosa.to_mono(audio.transpose(1, 0))
if sampling_rate != hps.data.sampling_rate:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=hps.data.sampling_rate)
with no_grad():
y = torch.FloatTensor(audio)
y = y.unsqueeze(0)
spec = spectrogram_torch(y, hps.data.filter_length,
hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length,
center=False)
spec_lengths = LongTensor([spec.size(-1)])
sid_src = LongTensor([original_speaker_id])
sid_tgt = LongTensor([target_speaker_id])
audio = model.voice_conversion(spec, spec_lengths, sid_src=sid_src, sid_tgt=sid_tgt)[0][
0, 0].data.cpu().float().numpy()
del y, spec, spec_lengths, sid_src, sid_tgt
return "Success", (hps.data.sampling_rate, audio)
return vc_fn
def create_soft_vc_fn(model, hps, speaker_ids):
def soft_vc_fn(target_speaker, input_audio1, input_audio2):
input_audio = input_audio1
if input_audio is None:
input_audio = input_audio2
if input_audio is None:
return "You need to upload an audio", None
sampling_rate, audio = input_audio
duration = audio.shape[0] / sampling_rate
if limitation and duration > 15:
return "Error: Audio is too long", None
target_speaker_id = speaker_ids[target_speaker]
audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
if len(audio.shape) > 1:
audio = librosa.to_mono(audio.transpose(1, 0))
if sampling_rate != 16000:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
with torch.inference_mode():
units = hubert.units(torch.FloatTensor(audio).unsqueeze(0).unsqueeze(0))
with no_grad():
unit_lengths = LongTensor([units.size(1)])
sid = LongTensor([target_speaker_id])
audio = model.infer(units, unit_lengths, sid=sid, noise_scale=.667,
noise_scale_w=0.8)[0][0, 0].data.cpu().float().numpy()
del units, unit_lengths, sid
return "Success", (hps.data.sampling_rate, audio)
return soft_vc_fn
def create_to_phoneme_fn(hps):
def to_phoneme_fn(text):
return _clean_text(text, hps.data.text_cleaners) if text != "" else ""
return to_phoneme_fn
css = """
#advanced-btn {
color: white;
border-color: black;
background: black;
font-size: .7rem !important;
line-height: 19px;
margin-top: 24px;
margin-bottom: 12px;
padding: 2px 8px;
border-radius: 14px !important;
}
#advanced-options {
display: none;
margin-bottom: 20px;
}
"""
if __name__ == '__main__':
models_tts = []
models_vc = []
models_soft_vc = []
with open("saved_model/info.json", "r", encoding="utf-8") as f:
models_info = json.load(f)
for i, info in models_info.items():
name = info["title"]
lang = info["lang"]
example = info["example"]
config_path = f"saved_model/{i}/config.json"
model_path = f"saved_model/{i}/model.pth"
cover_path = f"saved_model/{i}/cover.jpg"
hps = utils.get_hparams_from_file(config_path)
model = SynthesizerTrn(
len(hps.symbols),
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**hps.model)
utils.load_checkpoint(model_path, model, None)
model.eval()
speaker_ids = [sid for sid, name in enumerate(hps.speakers) if name != "None"]
speakers = [name for sid, name in enumerate(hps.speakers) if name != "None"]
t = info["type"]
if t == "vits":
models_tts.append((name, cover_path, speakers, lang, example,
hps.symbols, create_tts_fn(model, hps, speaker_ids),
create_to_phoneme_fn(hps)))
models_vc.append((name, cover_path, speakers, create_vc_fn(model, hps, speaker_ids)))
elif t == "soft-vits-vc":
models_soft_vc.append((name, cover_path, speakers, create_soft_vc_fn(model, hps, speaker_ids)))
hubert = torch.hub.load("bshall/hubert:main", "hubert_soft")
app = gr.Blocks(css=css)
with app:
gr.Markdown("# Moe TTS And Voice Conversion Using VITS Model\n\n"
"![visitor badge](https://visitor-badge.glitch.me/badge?page_id=skytnt.moegoe)\n\n"
"unofficial demo for \n\n"
"- [https://github.com/CjangCjengh/MoeGoe](https://github.com/CjangCjengh/MoeGoe)\n"
"- [https://github.com/Francis-Komizu/VITS](https://github.com/Francis-Komizu/VITS)\n"
"- [https://github.com/luoyily/MoeTTS](https://github.com/luoyily/MoeTTS)\n"
"- [https://github.com/Francis-Komizu/Sovits](https://github.com/Francis-Komizu/Sovits)"
)
with gr.Tabs():
with gr.TabItem("TTS"):
with gr.Tabs():
for i, (name, cover_path, speakers, lang, example, symbols, tts_fn,
to_phoneme_fn) in enumerate(models_tts):
with gr.TabItem(f"model{i}"):
with gr.Column():
gr.Markdown(f"## {name}\n\n"
f"![cover](file/{cover_path})\n\n"
f"lang: {lang}")
tts_input1 = gr.TextArea(label="Text (60 words limitation)", value=example,
elem_id=f"tts-input{i}")
tts_input2 = gr.Dropdown(label="Speaker", choices=speakers,
type="index", value=speakers[0])
tts_input3 = gr.Slider(label="Speed", value=1, minimum=0.5, maximum=2, step=0.1)
with gr.Accordion(label="Advanced Options", open=False):
phoneme_input = gr.Checkbox(value=False, label="Phoneme input")
to_phoneme_btn = gr.Button("Covert text to phoneme")
phoneme_list = gr.Dataset(label="Phoneme list", components=[tts_input1],
samples=[[x] for x in symbols])
phoneme_list_json = gr.Json(value=symbols, visible=False)
tts_submit = gr.Button("Generate", variant="primary")
tts_output1 = gr.Textbox(label="Output Message")
tts_output2 = gr.Audio(label="Output Audio")
tts_submit.click(tts_fn, [tts_input1, tts_input2, tts_input3, phoneme_input],
[tts_output1, tts_output2])
to_phoneme_btn.click(to_phoneme_fn, [tts_input1], [tts_input1])
phoneme_list.click(None, [phoneme_list, phoneme_list_json], [],
_js=f"""
(i,phonemes) => {{
let text_input = document.querySelector("body > gradio-app");
if (text_input.shadowRoot != null)
text_input = text_input.shadowRoot;
text_input = text_input.querySelector("#tts-input{i}").querySelector("textarea");
let startPos = text_input.selectionStart;
let endPos = text_input.selectionEnd;
let oldTxt = text_input.value;
let result = oldTxt.substring(0, startPos) + phonemes[i] + oldTxt.substring(endPos);
text_input.value = result;
text_input.focus()
text_input.selectionStart = startPos + phonemes[i].length;
text_input.selectionEnd = startPos + phonemes[i].length;
text_input.blur()
}}""")
with gr.TabItem("Voice Conversion"):
with gr.Tabs():
for i, (name, cover_path, speakers, vc_fn) in enumerate(models_vc):
with gr.TabItem(f"model{i}"):
gr.Markdown(f"## {name}\n\n"
f"![cover](file/{cover_path})")
vc_input1 = gr.Dropdown(label="Original Speaker", choices=speakers, type="index",
value=speakers[0])
vc_input2 = gr.Dropdown(label="Target Speaker", choices=speakers, type="index",
value=speakers[1])
vc_input3 = gr.Audio(label="Input Audio (15s limitation)")
vc_submit = gr.Button("Convert", variant="primary")
vc_output1 = gr.Textbox(label="Output Message")
vc_output2 = gr.Audio(label="Output Audio")
vc_submit.click(vc_fn, [vc_input1, vc_input2, vc_input3], [vc_output1, vc_output2])
with gr.TabItem("Soft Voice Conversion"):
with gr.Tabs():
for i, (name, cover_path, speakers, soft_vc_fn) in enumerate(models_soft_vc):
with gr.TabItem(f"model{i}"):
gr.Markdown(f"## {name}\n\n"
f"![cover](file/{cover_path})")
vc_input1 = gr.Dropdown(label="Target Speaker", choices=speakers, type="index",
value=speakers[0])
source_tabs = gr.Tabs()
with source_tabs:
with gr.TabItem("microphone"):
vc_input2 = gr.Audio(label="Input Audio (15s limitation)", source="microphone")
with gr.TabItem("upload"):
vc_input3 = gr.Audio(label="Input Audio (15s limitation)", source="upload")
vc_submit = gr.Button("Convert", variant="primary")
vc_output1 = gr.Textbox(label="Output Message")
vc_output2 = gr.Audio(label="Output Audio")
# clear inputs
source_tabs.set_event_trigger("change", None, [], [vc_input2, vc_input3],
js="()=>[null,null]")
vc_submit.click(soft_vc_fn, [vc_input1, vc_input2, vc_input3],
[vc_output1, vc_output2])
app.queue(concurrency_count=3).launch(show_api=False)