tts / voice_processing.py
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import asyncio
import datetime
import logging
import os
import time
import traceback
import tempfile
from concurrent.futures import ThreadPoolExecutor
import edge_tts
import librosa
import torch
from fairseq import checkpoint_utils
import uuid
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
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 get_unique_filename(extension):
return f"{uuid.uuid4()}.{extension}"
def model_data(model_name):
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", weights_only=True)
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)
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():
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 get_model_names():
return [d for d in os.listdir(model_root) if os.path.isdir(f"{model_root}/{d}")]
import nest_asyncio
nest_asyncio.apply()
def get_unique_filename(extension):
return f"{uuid.uuid4().hex[:8]}.{extension}"
class TTSProcessor:
def __init__(self, config):
self.config = config
self.executor = ThreadPoolExecutor(max_workers=config.n_cpu)
self.semaphore = asyncio.Semaphore(config.max_concurrent_tts)
self.last_request_time = time.time()
self.rate_limit = config.tts_rate_limit
self.temp_dir = tempfile.mkdtemp()
async def tts(self, model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice):
async with self.semaphore:
current_time = time.time()
time_since_last_request = current_time - self.last_request_time
if time_since_last_request < 1 / self.rate_limit:
await asyncio.sleep(1 / self.rate_limit - time_since_last_request)
self.last_request_time = time.time()
loop = asyncio.get_running_loop()
return await loop.run_in_executor(
self.executor,
self._tts_process,
model_name,
tts_text,
tts_voice,
index_rate,
use_uploaded_voice,
uploaded_voice
)
def _tts_process(self, model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice):
try:
edge_output_filename = os.path.join(self.temp_dir, get_unique_filename("mp3"))
if use_uploaded_voice:
if uploaded_voice is None:
return "No voice file uploaded.", None, None
uploaded_file_path = os.path.join(self.temp_dir, get_unique_filename("wav"))
with open(uploaded_file_path, "wb") as f:
f.write(uploaded_voice)
audio, sr = librosa.load(uploaded_file_path, sr=16000, mono=True)
else:
if limitation and len(tts_text) > 12000:
return (
f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.",
None,
None,
)
speed = 0 # Default speech speed
speed_str = f"+{speed}%" if speed >= 0 else f"{speed}%"
# Use synchronous approach for Edge TTS
communicate = edge_tts.Communicate(tts_text, tts_voice, rate=speed_str)
asyncio.get_event_loop().run_until_complete(communicate.save(edge_output_filename))
audio, sr = librosa.load(edge_output_filename, sr=16000, mono=True)
duration = len(audio) / sr
if limitation and duration >= 20000:
return (
f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
None,
None,
)
f0_up_key = 0
tgt_sr, net_g, vc, version, index_file, if_f0 = model_data(model_name)
if hasattr(self, 'model_rmvpe'):
vc.model_rmvpe = self.model_rmvpe
times = [0, 0, 0]
audio_opt = vc.pipeline(
hubert_model,
net_g,
0,
audio,
edge_output_filename if not use_uploaded_voice else uploaded_file_path,
times,
f0_up_key,
"rmvpe",
index_file,
index_rate,
if_f0,
3, # filter_radius
tgt_sr,
0, # resample_sr
0.25, # rms_mix_rate
version,
0.33, # protect
None,
)
info = f"Success. Time: tts: {times[0]}s, npy: {times[1]}s, f0: {times[2]}s"
print(info)
return (
info,
edge_output_filename if not use_uploaded_voice else None,
(tgt_sr, audio_opt),
)
except Exception as e:
logging.error(f"Error in TTS processing: {str(e)}")
logging.error(traceback.format_exc())
return str(e), None, None
def __del__(self):
# Clean up temporary directory
import shutil
shutil.rmtree(self.temp_dir, ignore_errors=True)
# Initialize global variables
tts_processor = TTSProcessor(config)
hubert_model = load_hubert()
rmvpe_model = RMVPE("rmvpe.pt", config.is_half, config.device)
tts_processor.model_rmvpe = rmvpe_model
voice_mapping = {
"Mongolian Male": "mn-MN-BataaNeural",
"Mongolian Female": "mn-MN-YesuiNeural"
}