TTS_Mongolian / app.py
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import asyncio
import datetime
import logging
import os
import time
import traceback
import edge_tts
import gradio as gr
import librosa
import torch
from fairseq import checkpoint_utils
from config import Config
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from rmvpe import RMVPE
from vc_infer_pipeline import VC
# Set logging levels
logging.getLogger("fairseq").setLevel(logging.WARNING)
logging.getLogger("numba").setLevel(logging.WARNING)
logging.getLogger("markdown_it").setLevel(logging.WARNING)
logging.getLogger("urllib3").setLevel(logging.WARNING)
logging.getLogger("matplotlib").setLevel(logging.WARNING)
limitation = os.getenv("SYSTEM") == "spaces"
config = Config()
# Edge TTS
edge_output_filename = "edge_output.mp3"
tts_voice_list = asyncio.get_event_loop().run_until_complete(edge_tts.list_voices())
tts_voices = ["mn-MN-BataaNeural", "mn-MN-YesuiNeural"] # Specific voices
# RVC models
model_root = "weights"
models = [d for d in os.listdir(model_root) if os.path.isdir(f"{model_root}/{d}")]
models.sort()
def model_data(model_name):
# global n_spk, tgt_sr, net_g, vc, cpt, version, index_file
pth_path = [
f"{model_root}/{model_name}/{f}"
for f in os.listdir(f"{model_root}/{model_name}")
if f.endswith(".pth")
][0]
print(f"Loading {pth_path}")
cpt = torch.load(pth_path, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
else:
raise ValueError("Unknown version")
del net_g.enc_q
net_g.load_state_dict(cpt["weight"], strict=False)
print("Model loaded")
net_g.eval().to(config.device)
if config.is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
# n_spk = cpt["config"][-3]
index_files = [
f"{model_root}/{model_name}/{f}"
for f in os.listdir(f"{model_root}/{model_name}")
if f.endswith(".index")
]
if len(index_files) == 0:
print("No index file found")
index_file = ""
else:
index_file = index_files[0]
print(f"Index file found: {index_file}")
return tgt_sr, net_g, vc, version, index_file, if_f0
def load_hubert():
# global hubert_model
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
if config.is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
return hubert_model.eval()
def tts(
model_name,
speed,
tts_text,
tts_voice,
f0_up_key,
f0_method,
index_rate,
protect,
filter_radius=3,
resample_sr=0,
rms_mix_rate=0.25,
):
print("------------------")
print(datetime.datetime.now())
print("tts_text:")
print(tts_text)
print(f"tts_voice: {tts_voice}, speed: {speed}")
print(f"Model name: {model_name}")
print(f"F0: {f0_method}, Key: {f0_up_key}, Index: {index_rate}, Protect: {protect}")
try:
if limitation and len(tts_text) > 280:
print("Error: Text too long")
return (
f"Text characters should be at most 280 in this huggingface space, but got {len(tts_text)} characters.",
None,
None,
)
t0 = time.time()
if speed >= 0:
speed_str = f"+{speed}%"
else:
speed_str = f"{speed}%"
asyncio.run(
edge_tts.Communicate(
tts_text, tts_voice, rate=speed_str
).save(edge_output_filename)
)
t1 = time.time()
edge_time = t1 - t0
audio, sr = librosa.load(edge_output_filename, sr=16000, mono=True)
duration = len(audio) / sr
print(f"Audio duration: {duration}s")
if limitation and duration >= 20:
print("Error: Audio too long")
return (
f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
edge_output_filename,
None,
)
f0_up_key = int(f0_up_key)
tgt_sr, net_g, vc, version, index_file, if_f0 = model_data(model_name)
if f0_method == "rmvpe":
vc.model_rmvpe = rmvpe_model
times = [0, 0, 0]
audio_opt = vc.pipeline(
hubert_model,
net_g,
0,
audio,
edge_output_filename,
times,
f0_up_key,
f0_method,
index_file,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
None,
)
if tgt_sr != resample_sr >= 16000:
tgt_sr = resample_sr
info = f"Success. Time: edge-tts: {edge_time}s, npy: {times[0]}s, f0: {times[1]}s, infer: {times[2]}s"
print(info)
return (
info,
edge_output_filename,
(tgt_sr, audio_opt),
)
except EOFError:
info = (
"It seems that the edge-tts output is not valid. "
"This may occur when the input text and the speaker do not match. "
"For example, maybe you entered Japanese (without alphabets) text but chose non-Japanese speaker?"
)
print(info)
return info, None, None
except:
info = traceback.format_exc()
print(info)
return info, None, None
print("Loading hubert model...")
hubert_model = load_hubert()
print("Hubert model loaded.")
print("Loading rmvpe model...")
rmvpe_model = RMVPE("rmvpe.pt", config.is_half, config.device)
print("rmvpe model loaded.")
initial_md = """
# RVC text-to-speech demo
This is a text-to-speech demo of RVC moe models of [rvc_okiba](https://huggingface.co/litagin/rvc_okiba) using [edge-tts](https://github.com/rany2/edge-tts).
Input text ➡[(edge-tts)](https://github.com/rany2/edge-tts)➡ Speech mp3 file ➡[(RVC)](https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI)➡ Final output
This runs on the 🤗 server's cpu, so it may be slow.
Although the models are trained on Japanese voices and intended for Japanese text, they can also be used with other languages with the corresponding edge-tts speaker (but possibly with a Japanese accent).
Input characters are limited to 280 characters, and the speech audio is limited to 20 seconds in this 🤗 space.
[Visit this GitHub repo](https://github.com/litagin02/rvc-tts-webui) for running locally with your models and GPU!
"""
app = gr.Blocks()
with app:
with gr.Row():
with gr.Column():
model_name = gr.Dropdown(
label="Model (all models except man-_ are girl models)",
choices=models,
value=models[0],
)
f0_key_up = gr.Number(
label="Tune (+12 = 1 octave up from edge-tts, the best value depends on the models and speakers)",
value=0,
)
with gr.Column():
f0_method = gr.Radio(
label="Pitch extraction method (pm: very fast, low quality, rmvpe: a little slow, high quality)",
choices=["pm", "rmvpe"],
value="rmvpe",
interactive=True,
)
index_rate = gr.Slider(
minimum=0,
maximum=1,
label="Slang rate",
value=0.75,
interactive=True,
)
protect0 = gr.Slider(
minimum=0,
maximum=0.5,
label="Protect",
value=0.33,
step=0.01,
interactive=True,
)
with gr.Row():
with gr.Column():
tts_voice = gr.Dropdown(
label="Edge-tts speaker (format: language-Country-Name-Gender)",
choices=tts_voices,
allow_custom_value=False,
value="mn-MN-BataaNeural",
)
speed = gr.Slider(
minimum=-100,
maximum=100,
label="Speech speed (%)",
value=0,
step=10,
interactive=True,
)
tts_text = gr.Textbox(label="Input Text", value="Текстыг оруулна уу.")
with gr.Column():
but0 = gr.Button("Convert", variant="primary")
info_text = gr.Textbox(label="Output info")
with gr.Column():
edge_tts_output = gr.Audio(label="Edge Voice", type="filepath")
tts_output = gr.Audio(label="Result")
but0.click(
tts,
[
model_name,
speed,
tts_text,
tts_voice,
f0_key_up,
f0_method,
index_rate,
protect0,
],
[info_text, edge_tts_output, tts_output],
)
app.launch()