moe-tts-tech / app.py
alixya's picture
Duplicate from AsakuraMizu/moe-tts
f7d71e2
import argparse
import json
import os
import re
import tempfile
from pathlib import Path
import librosa
import numpy as np
import torch
from torch import no_grad, LongTensor
import commons
import utils
import gradio as gr
import gradio.utils as gr_utils
import gradio.processing_utils as gr_processing_utils
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
audio_postprocess_ori = gr.Audio.postprocess
def audio_postprocess(self, y):
data = audio_postprocess_ori(self, y)
if data is None:
return None
return gr_processing_utils.encode_url_or_file_to_base64(data["name"])
gr.Audio.postprocess = audio_postprocess
def get_text(text, hps, is_symbol):
text_norm = text_to_sequence(text, hps.symbols, [] if is_symbol 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_symbol):
if limitation:
text_len = len(re.sub("\[([A-Z]{2})\]", "", text))
max_len = 150
if is_symbol:
max_len *= 3
if text_len > max_len:
return "Error: Text is too long", None
speaker_id = speaker_ids[speaker]
stn_tst = get_text(text, hps, is_symbol)
with no_grad():
x_tst = stn_tst.unsqueeze(0).to(device)
x_tst_lengths = LongTensor([stn_tst.size(0)]).to(device)
sid = LongTensor([speaker_id]).to(device)
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 > 30:
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).to(device)
spec_lengths = LongTensor([spec.size(-1)]).to(device)
sid_src = LongTensor([original_speaker_id]).to(device)
sid_tgt = LongTensor([target_speaker_id]).to(device)
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 > 30:
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).to(device))
with no_grad():
unit_lengths = LongTensor([units.size(1)]).to(device)
sid = LongTensor([target_speaker_id]).to(device)
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_symbol_fn(hps):
def to_symbol_fn(is_symbol_input, input_text, temp_text):
return (_clean_text(input_text, hps.data.text_cleaners), input_text) if is_symbol_input \
else (temp_text, temp_text)
return to_symbol_fn
download_audio_js = """
() =>{{
let root = document.querySelector("body > gradio-app");
if (root.shadowRoot != null)
root = root.shadowRoot;
let audio = root.querySelector("#{audio_id}").querySelector("audio");
if (audio == undefined)
return;
audio = audio.src;
let oA = document.createElement("a");
oA.download = Math.floor(Math.random()*100000000)+'.wav';
oA.href = audio;
document.body.appendChild(oA);
oA.click();
oA.remove();
}}
"""
if __name__ == '__main__':
parser = argparse.ArgumentParser()
parser.add_argument('--device', type=str, default='cpu')
parser.add_argument("--share", action="store_true", default=False, help="share gradio app")
args = parser.parse_args()
device = torch.device(args.device)
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"]
author = info["author"]
lang = info["lang"]
example = info["example"]
config_path = f"saved_model/{i}/config.json"
model_path = f"saved_model/{i}/model.pth"
cover = info["cover"]
cover_path = f"saved_model/{i}/{cover}" if cover else None
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().to(device)
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, author, cover_path, speakers, lang, example,
hps.symbols, create_tts_fn(model, hps, speaker_ids),
create_to_symbol_fn(hps)))
models_vc.append((name, author, cover_path, speakers, create_vc_fn(model, hps, speaker_ids)))
elif t == "soft-vits-vc":
models_soft_vc.append((name, author, cover_path, speakers, create_soft_vc_fn(model, hps, speaker_ids)))
hubert = torch.hub.load("bshall/hubert:main", "hubert_soft", trust_repo=True).to(device)
app = gr.Blocks()
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"
"[Open In Colab]"
"(https://colab.research.google.com/drive/14Pb8lpmwZL-JI5Ub6jpG4sz2-8KS0kbS?usp=sharing)"
" without queue and length limitation.\n\n"
"Feel free to [open discussion](https://huggingface.co/spaces/skytnt/moe-tts/discussions/new) "
"if you want to add your model to this app.")
with gr.Tabs():
with gr.TabItem("TTS"):
with gr.Tabs():
for i, (name, author, cover_path, speakers, lang, example, symbols, tts_fn,
to_symbol_fn) in enumerate(models_tts):
with gr.TabItem(f"model{i}"):
with gr.Column():
cover_markdown = f"![cover](file/{cover_path})\n\n" if cover_path else ""
gr.Markdown(f"## {name}\n\n"
f"{cover_markdown}"
f"model author: {author}\n\n"
f"language: {lang}")
tts_input1 = gr.TextArea(label="Text (150 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):
temp_text_var = gr.Variable()
symbol_input = gr.Checkbox(value=False, label="Symbol input")
symbol_list = gr.Dataset(label="Symbol list", components=[tts_input1],
samples=[[x] for x in symbols],
elem_id=f"symbol-list{i}")
symbol_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", elem_id=f"tts-audio{i}")
download = gr.Button("Download Audio")
download.click(None, [], [], _js=download_audio_js.format(audio_id=f"tts-audio{i}"))
tts_submit.click(tts_fn, [tts_input1, tts_input2, tts_input3, symbol_input],
[tts_output1, tts_output2], api_name=f"tts-model{i}")
symbol_input.change(to_symbol_fn,
[symbol_input, tts_input1, temp_text_var],
[tts_input1, temp_text_var])
symbol_list.click(None, [symbol_list, symbol_list_json], [],
_js=f"""
(i,symbols) => {{
let root = document.querySelector("body > gradio-app");
if (root.shadowRoot != null)
root = root.shadowRoot;
let text_input = root.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) + symbols[i] + oldTxt.substring(endPos);
text_input.value = result;
let x = window.scrollX, y = window.scrollY;
text_input.focus();
text_input.selectionStart = startPos + symbols[i].length;
text_input.selectionEnd = startPos + symbols[i].length;
text_input.blur();
window.scrollTo(x, y);
return [];
}}""")
with gr.TabItem("Voice Conversion"):
with gr.Tabs():
for i, (name, author, cover_path, speakers, vc_fn) in enumerate(models_vc):
with gr.TabItem(f"model{i}"):
cover_markdown = f"![cover](file/{cover_path})\n\n" if cover_path else ""
gr.Markdown(f"## {name}\n\n"
f"{cover_markdown}"
f"model author: {author}")
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[min(len(speakers) - 1, 1)])
vc_input3 = gr.Audio(label="Input Audio (30s limitation)")
vc_submit = gr.Button("Convert", variant="primary")
vc_output1 = gr.Textbox(label="Output Message")
vc_output2 = gr.Audio(label="Output Audio", elem_id=f"vc-audio{i}")
download = gr.Button("Download Audio")
download.click(None, [], [], _js=download_audio_js.format(audio_id=f"vc-audio{i}"))
vc_submit.click(vc_fn, [vc_input1, vc_input2, vc_input3], [vc_output1, vc_output2], api_name=f"vc-model{i}")
with gr.TabItem("Soft Voice Conversion"):
with gr.Tabs():
for i, (name, author, cover_path, speakers, soft_vc_fn) in enumerate(models_soft_vc):
with gr.TabItem(f"model{i}"):
cover_markdown = f"![cover](file/{cover_path})\n\n" if cover_path else ""
gr.Markdown(f"## {name}\n\n"
f"{cover_markdown}"
f"model author: {author}")
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 (30s limitation)", source="microphone")
with gr.TabItem("upload"):
vc_input3 = gr.Audio(label="Input Audio (30s limitation)", source="upload")
vc_submit = gr.Button("Convert", variant="primary")
vc_output1 = gr.Textbox(label="Output Message")
vc_output2 = gr.Audio(label="Output Audio", elem_id=f"svc-audio{i}")
download = gr.Button("Download Audio")
download.click(None, [], [], _js=download_audio_js.format(audio_id=f"svc-audio{i}"))
# 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], api_name=f"svc-model{i}")
gr.Markdown(
"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)"
)
app.queue(concurrency_count=3).launch(share=args.share)