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 base64
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_voices():
return list(voice_mapping.keys())
def get_model_names():
model_root = "weights" # Adjust this path if your models are stored elsewhere
return [d for d in os.listdir(model_root) if os.path.isdir(f"{model_root}/{d}")]
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")
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()
# Add this helper function to ensure a new event loop is created if none exists
def run_async_in_thread(fn, *args):
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
result = loop.run_until_complete(fn(*args))
loop.close()
return result
def parallel_tts(tasks):
with ThreadPoolExecutor() as executor:
futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks]
results = [future.result() for future in futures]
return results
async def tts(
model_name,
tts_text,
tts_voice,
index_rate,
use_uploaded_voice,
uploaded_voice,
):
edge_output_filename = get_unique_filename("mp3")
try:
# Default values for parameters
speed = 0
f0_up_key = 0
f0_method = "rmvpe"
protect = 0.33
filter_radius = 3
resample_sr = 0
rms_mix_rate = 0.25
edge_time = 0
if use_uploaded_voice:
if uploaded_voice is None:
raise ValueError("No voice file uploaded.")
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp_file:
tmp_file.write(uploaded_voice)
uploaded_file_path = tmp_file.name
audio, sr = librosa.load(uploaded_file_path, sr=16000, mono=True)
else:
if limitation and len(tts_text) > 12000:
raise ValueError(f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.")
t0 = time.time()
speed_str = f"+{speed}%" if speed >= 0 else f"{speed}%"
await edge_tts.Communicate(tts_text, tts_voice, rate=speed_str).save(edge_output_filename)
edge_time = time.time() - 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 >= 20000:
raise ValueError(f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.")
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 if not use_uploaded_voice else uploaded_file_path,
times,
f0_up_key,
f0_method,
index_file,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
None,
)
if tgt_sr != resample_sr and resample_sr >= 16000:
tgt_sr = resample_sr
info = f"Success. Time: tts: {edge_time:.2f}s, npy: {times[0]:.2f}s, f0: {times[1]:.2f}s, infer: {times[2]:.2f}s"
print(info)
# Convert audio to base64
with open(edge_output_filename, "rb") as audio_file:
audio_base64 = base64.b64encode(audio_file.read()).decode('utf-8')
audio_data_uri = f"data:audio/mp3;base64,{audio_base64}"
return (
info,
audio_data_uri,
(tgt_sr, audio_opt) # Return the target sample rate and audio output
)
except Exception as e:
print(f"Error in TTS task: {str(e)}")
import traceback
traceback.print_exc()
if os.path.exists(edge_output_filename):
os.remove(edge_output_filename)
return (str(e), None, None)
voice_mapping = {
"Mongolian Male": "mn-MN-BataaNeural",
"Mongolian Female": "mn-MN-YesuiNeural"
# Add more mappings as needed
}
hubert_model = load_hubert()
rmvpe_model = RMVPE("rmvpe.pt", config.is_half, config.device)
# Global semaphore to control concurrency
max_concurrent_tasks = 16 # Adjust based on server capacity
semaphore = asyncio.Semaphore(max_concurrent_tasks)
# Global ThreadPoolExecutor
executor = ThreadPoolExecutor(max_workers=max_concurrent_tasks)
class TTSProcessor:
def __init__(self, config):
self.config = config
self.queue = asyncio.Queue()
self.is_processing = False
async def tts(self, model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice):
task = asyncio.create_task(self._tts_task(model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice))
await self.queue.put(task)
if not self.is_processing:
asyncio.create_task(self._process_queue())
return await task
async def _tts_task(self, model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice):
async with semaphore:
return await tts(model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice)
async def _process_queue(self):
self.is_processing = True
while not self.queue.empty():
task = await self.queue.get()
try:
await task
except asyncio.CancelledError:
print("Task was cancelled")
except Exception as e:
print(f"Task failed with error: {e}")
finally:
self.queue.task_done()
self.is_processing = False
# Initialize the TTSProcessor
tts_processor = TTSProcessor(config)
async def parallel_tts_processor(tasks):
return await asyncio.gather(*(tts_processor.tts(*task) for task in tasks))
async def parallel_tts_wrapper(tasks):
return await parallel_tts_processor(tasks)