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import asyncio |
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import datetime |
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import logging |
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import os |
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import time |
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import traceback |
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import tempfile |
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from concurrent.futures import ThreadPoolExecutor |
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import base64 |
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import edge_tts |
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import librosa |
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import torch |
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from fairseq import checkpoint_utils |
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import uuid |
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from config import Config |
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from lib.infer_pack.models import ( |
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SynthesizerTrnMs256NSFsid, |
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SynthesizerTrnMs256NSFsid_nono, |
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SynthesizerTrnMs768NSFsid, |
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SynthesizerTrnMs768NSFsid_nono, |
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) |
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from rmvpe import RMVPE |
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from vc_infer_pipeline import VC |
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logging.getLogger("fairseq").setLevel(logging.WARNING) |
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logging.getLogger("numba").setLevel(logging.WARNING) |
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logging.getLogger("markdown_it").setLevel(logging.WARNING) |
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logging.getLogger("urllib3").setLevel(logging.WARNING) |
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logging.getLogger("matplotlib").setLevel(logging.WARNING) |
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limitation = os.getenv("SYSTEM") == "spaces" |
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config = Config() |
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tts_voice_list = asyncio.get_event_loop().run_until_complete(edge_tts.list_voices()) |
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tts_voices = ["mn-MN-BataaNeural", "mn-MN-YesuiNeural"] |
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model_root = "weights" |
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models = [d for d in os.listdir(model_root) if os.path.isdir(f"{model_root}/{d}")] |
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models.sort() |
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def get_voices(): |
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return list(voice_mapping.keys()) |
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def get_model_names(): |
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model_root = "weights" |
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return [d for d in os.listdir(model_root) if os.path.isdir(f"{model_root}/{d}")] |
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def get_unique_filename(extension): |
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return f"{uuid.uuid4()}.{extension}" |
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def model_data(model_name): |
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pth_path = [ |
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f"{model_root}/{model_name}/{f}" |
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for f in os.listdir(f"{model_root}/{model_name}") |
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if f.endswith(".pth") |
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][0] |
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print(f"Loading {pth_path}") |
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cpt = torch.load(pth_path, map_location="cpu") |
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tgt_sr = cpt["config"][-1] |
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cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] |
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if_f0 = cpt.get("f0", 1) |
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version = cpt.get("version", "v1") |
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if version == "v1": |
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if if_f0 == 1: |
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net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half) |
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else: |
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net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"]) |
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elif version == "v2": |
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if if_f0 == 1: |
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net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half) |
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else: |
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net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"]) |
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else: |
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raise ValueError("Unknown version") |
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del net_g.enc_q |
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net_g.load_state_dict(cpt["weight"], strict=False) |
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print("Model loaded") |
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net_g.eval().to(config.device) |
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if config.is_half: |
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net_g = net_g.half() |
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else: |
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net_g = net_g.float() |
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vc = VC(tgt_sr, config) |
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index_files = [ |
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f"{model_root}/{model_name}/{f}" |
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for f in os.listdir(f"{model_root}/{model_name}") |
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if f.endswith(".index") |
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] |
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if len(index_files) == 0: |
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print("No index file found") |
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index_file = "" |
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else: |
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index_file = index_files[0] |
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print(f"Index file found: {index_file}") |
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return tgt_sr, net_g, vc, version, index_file, if_f0 |
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def load_hubert(): |
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models, _, _ = checkpoint_utils.load_model_ensemble_and_task( |
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["hubert_base.pt"], |
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suffix="", |
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) |
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hubert_model = models[0] |
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hubert_model = hubert_model.to(config.device) |
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if config.is_half: |
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hubert_model = hubert_model.half() |
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else: |
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hubert_model = hubert_model.float() |
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return hubert_model.eval() |
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def run_async_in_thread(fn, *args): |
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loop = asyncio.new_event_loop() |
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asyncio.set_event_loop(loop) |
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result = loop.run_until_complete(fn(*args)) |
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loop.close() |
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return result |
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def parallel_tts(tasks): |
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with ThreadPoolExecutor() as executor: |
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futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks] |
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results = [future.result() for future in futures] |
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return results |
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async def tts( |
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model_name, |
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tts_text, |
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tts_voice, |
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index_rate, |
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use_uploaded_voice, |
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uploaded_voice, |
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): |
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edge_output_filename = get_unique_filename("mp3") |
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try: |
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speed = 0 |
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f0_up_key = 0 |
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f0_method = "rmvpe" |
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protect = 0.33 |
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filter_radius = 3 |
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resample_sr = 0 |
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rms_mix_rate = 0.25 |
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edge_time = 0 |
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if use_uploaded_voice: |
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if uploaded_voice is None: |
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raise ValueError("No voice file uploaded.") |
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with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp_file: |
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tmp_file.write(uploaded_voice) |
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uploaded_file_path = tmp_file.name |
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audio, sr = librosa.load(uploaded_file_path, sr=16000, mono=True) |
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else: |
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if limitation and len(tts_text) > 12000: |
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raise ValueError(f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.") |
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t0 = time.time() |
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speed_str = f"+{speed}%" if speed >= 0 else f"{speed}%" |
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await edge_tts.Communicate(tts_text, tts_voice, rate=speed_str).save(edge_output_filename) |
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edge_time = time.time() - t0 |
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audio, sr = librosa.load(edge_output_filename, sr=16000, mono=True) |
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duration = len(audio) / sr |
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print(f"Audio duration: {duration}s") |
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if limitation and duration >= 20000: |
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raise ValueError(f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.") |
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f0_up_key = int(f0_up_key) |
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tgt_sr, net_g, vc, version, index_file, if_f0 = model_data(model_name) |
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if f0_method == "rmvpe": |
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vc.model_rmvpe = rmvpe_model |
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times = [0, 0, 0] |
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audio_opt = vc.pipeline( |
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hubert_model, |
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net_g, |
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0, |
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audio, |
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edge_output_filename if not use_uploaded_voice else uploaded_file_path, |
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times, |
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f0_up_key, |
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f0_method, |
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index_file, |
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index_rate, |
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if_f0, |
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filter_radius, |
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tgt_sr, |
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resample_sr, |
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rms_mix_rate, |
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version, |
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protect, |
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None, |
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) |
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if tgt_sr != resample_sr and resample_sr >= 16000: |
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tgt_sr = resample_sr |
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info = f"Success. Time: tts: {edge_time:.2f}s, npy: {times[0]:.2f}s, f0: {times[1]:.2f}s, infer: {times[2]:.2f}s" |
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print(info) |
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with open(edge_output_filename, "rb") as audio_file: |
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audio_base64 = base64.b64encode(audio_file.read()).decode('utf-8') |
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audio_data_uri = f"data:audio/mp3;base64,{audio_base64}" |
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return ( |
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info, |
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audio_data_uri, |
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(tgt_sr, audio_opt) |
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) |
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except Exception as e: |
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print(f"Error in TTS task: {str(e)}") |
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import traceback |
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traceback.print_exc() |
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if os.path.exists(edge_output_filename): |
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os.remove(edge_output_filename) |
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return (str(e), None, None) |
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voice_mapping = { |
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"Mongolian Male": "mn-MN-BataaNeural", |
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"Mongolian Female": "mn-MN-YesuiNeural" |
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} |
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hubert_model = load_hubert() |
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rmvpe_model = RMVPE("rmvpe.pt", config.is_half, config.device) |
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max_concurrent_tasks = 16 |
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semaphore = asyncio.Semaphore(max_concurrent_tasks) |
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executor = ThreadPoolExecutor(max_workers=max_concurrent_tasks) |
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class TTSProcessor: |
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def __init__(self, config): |
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self.config = config |
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self.queue = asyncio.Queue() |
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self.is_processing = False |
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async def tts(self, model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice): |
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task = asyncio.create_task(self._tts_task(model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice)) |
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await self.queue.put(task) |
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if not self.is_processing: |
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asyncio.create_task(self._process_queue()) |
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return await task |
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async def _tts_task(self, model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice): |
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async with semaphore: |
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return await tts(model_name, tts_text, tts_voice, index_rate, use_uploaded_voice, uploaded_voice) |
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async def _process_queue(self): |
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self.is_processing = True |
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while not self.queue.empty(): |
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task = await self.queue.get() |
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try: |
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await task |
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except asyncio.CancelledError: |
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print("Task was cancelled") |
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except Exception as e: |
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print(f"Task failed with error: {e}") |
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finally: |
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self.queue.task_done() |
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self.is_processing = False |
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tts_processor = TTSProcessor(config) |
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async def parallel_tts_processor(tasks): |
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return await asyncio.gather(*(tts_processor.tts(*task) for task in tasks)) |
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async def parallel_tts_wrapper(tasks): |
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return await parallel_tts_processor(tasks) |