Create api.py
Create the initial API Python file for easy expansion and use.
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api.py
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402
api.py
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import os
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import sys
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import json
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import re
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import time
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import librosa
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import torch
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import numpy as np
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import torch.nn.functional as F
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import torchaudio.transforms as tat
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import sounddevice as sd
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from dotenv import load_dotenv
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from fastapi import FastAPI, HTTPException
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from pydantic import BaseModel
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import threading
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import uvicorn
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from tools.torchgate import TorchGate
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import tools.rvc_for_realtime as rvc_for_realtime
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from configs.config import Config
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load_dotenv()
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os.environ["OMP_NUM_THREADS"] = "4"
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if sys.platform == "darwin":
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os.environ["PYTORCH_ENABLE_MPS_FALLBACK"] = "1"
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now_dir = os.getcwd()
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sys.path.append(now_dir)
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stream_latency = -1
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app = FastAPI()
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class GUIConfig:
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def __init__(self) -> None:
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self.pth_path: str = ""
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self.index_path: str = ""
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self.pitch: int = 0
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self.samplerate: int = 40000
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self.block_time: float = 1.0 # s
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self.buffer_num: int = 1
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self.threhold: int = -60
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self.crossfade_time: float = 0.05
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self.extra_time: float = 2.5
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self.I_noise_reduce = False
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self.O_noise_reduce = False
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self.rms_mix_rate = 0.0
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self.index_rate = 0.3
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self.f0method = "rmvpe"
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self.sg_input_device = ""
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self.sg_output_device = ""
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class ConfigData(BaseModel):
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pth_path: str
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index_path: str
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sg_input_device: str
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sg_output_device: str
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threhold: int = -60
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pitch: int = 0
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index_rate: float = 0.3
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rms_mix_rate: float = 0.0
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block_time: float = 0.25
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crossfade_length: float = 0.05
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extra_time: float = 2.5
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n_cpu: int = 4
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I_noise_reduce: bool = False
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O_noise_reduce: bool = False
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class AudioAPI:
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def __init__(self) -> None:
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self.gui_config = GUIConfig()
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self.config = Config()
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self.flag_vc = False
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self.function = "vc"
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self.delay_time = 0
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def load(self):
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input_devices, output_devices, _, _ = self.get_devices()
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try:
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with open("configs/config.json", "r") as j:
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data = json.load(j)
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data["rmvpe"] = True # Ensure rmvpe is the only f0method
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if data["sg_input_device"] not in input_devices:
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data["sg_input_device"] = input_devices[sd.default.device[0]]
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if data["sg_output_device"] not in output_devices:
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data["sg_output_device"] = output_devices[sd.default.device[1]]
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except:
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with open("configs/config.json", "w") as j:
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data = {
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"pth_path": " ",
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"index_path": " ",
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"sg_input_device": input_devices[sd.default.device[0]],
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"sg_output_device": output_devices[sd.default.device[1]],
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"threhold": "-60",
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"pitch": "0",
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"index_rate": "0",
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"rms_mix_rate": "0",
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"block_time": "0.25",
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"crossfade_length": "0.05",
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"extra_time": "2.5",
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"f0method": "rmvpe",
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"use_jit": False,
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}
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data["rmvpe"] = True # Ensure rmvpe is the only f0method
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return data
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def set_values(self, values):
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if len(values.pth_path.strip()) == 0:
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raise HTTPException(status_code=400, detail="Please select a .pth file")
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if len(values.index_path.strip()) == 0:
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raise HTTPException(status_code=400, detail="Please select an index file")
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pattern = re.compile("[^\x00-\x7F]+")
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if pattern.findall(values.pth_path):
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raise HTTPException(status_code=400, detail=".pth file path should not contain non-ASCII characters")
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if pattern.findall(values.index_path):
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raise HTTPException(status_code=400, detail="Index file path should not contain non-ASCII characters")
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self.set_devices(values.sg_input_device, values.sg_output_device)
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self.config.use_jit = False
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self.gui_config.pth_path = values.pth_path
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self.gui_config.index_path = values.index_path
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self.gui_config.threhold = values.threhold
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self.gui_config.pitch = values.pitch
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self.gui_config.block_time = values.block_time
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self.gui_config.crossfade_time = values.crossfade_length
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self.gui_config.extra_time = values.extra_time
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self.gui_config.I_noise_reduce = values.I_noise_reduce
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self.gui_config.O_noise_reduce = values.O_noise_reduce
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self.gui_config.rms_mix_rate = values.rms_mix_rate
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self.gui_config.index_rate = values.index_rate
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self.gui_config.n_cpu = values.n_cpu
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self.gui_config.f0method = "rmvpe"
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return True
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def start_vc(self):
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torch.cuda.empty_cache()
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self.flag_vc = True
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self.rvc = rvc_for_realtime.RVC(
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self.gui_config.pitch,
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self.gui_config.pth_path,
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self.gui_config.index_path,
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self.gui_config.index_rate,
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0,
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0,
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0,
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self.config,
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self.rvc if hasattr(self, "rvc") else None,
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)
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if not hasattr(self.rvc, 'tgt_sr'):
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self.rvc.tgt_sr = 44100
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self.gui_config.samplerate = self.rvc.tgt_sr
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self.zc = self.rvc.tgt_sr // 100
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self.block_frame = (
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int(
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np.round(
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self.gui_config.block_time
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* self.gui_config.samplerate
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/ self.zc
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)
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)
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* self.zc
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)
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self.block_frame_16k = 160 * self.block_frame // self.zc
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self.crossfade_frame = (
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int(
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np.round(
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self.gui_config.crossfade_time
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* self.gui_config.samplerate
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/ self.zc
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)
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)
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* self.zc
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)
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self.sola_search_frame = self.zc
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self.extra_frame = (
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int(
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np.round(
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self.gui_config.extra_time
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* self.gui_config.samplerate
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/ self.zc
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)
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)
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* self.zc
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)
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self.input_wav = torch.zeros(
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self.extra_frame + self.crossfade_frame + self.sola_search_frame + self.block_frame,
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device=self.config.device,
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dtype=torch.float32,
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)
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self.input_wav_res = torch.zeros(
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160 * self.input_wav.shape[0] // self.zc,
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device=self.config.device,
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dtype=torch.float32,
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)
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self.pitch = np.zeros(self.input_wav.shape[0] // self.zc, dtype="int32")
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self.pitchf = np.zeros(self.input_wav.shape[0] // self.zc, dtype="float64")
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self.sola_buffer = torch.zeros(self.crossfade_frame, device=self.config.device, dtype=torch.float32)
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self.nr_buffer = self.sola_buffer.clone()
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self.output_buffer = self.input_wav.clone()
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self.res_buffer = torch.zeros(2 * self.zc, device=self.config.device, dtype=torch.float32)
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self.valid_rate = 1 - (self.extra_frame - 1) / self.input_wav.shape[0]
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self.fade_in_window = (
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torch.sin(0.5 * np.pi * torch.linspace(0.0, 1.0, steps=self.crossfade_frame, device=self.config.device, dtype=torch.float32)) ** 2
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)
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self.fade_out_window = 1 - self.fade_in_window
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self.resampler = tat.Resample(
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orig_freq=self.gui_config.samplerate,
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new_freq=16000,
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dtype=torch.float32,
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).to(self.config.device)
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self.tg = TorchGate(
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sr=self.gui_config.samplerate, n_fft=4 * self.zc, prop_decrease=0.9
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).to(self.config.device)
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thread_vc = threading.Thread(target=self.soundinput)
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thread_vc.start()
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def soundinput(self):
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channels = 1 if sys.platform == "darwin" else 2
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with sd.Stream(
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channels=channels,
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callback=self.audio_callback,
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blocksize=self.block_frame,
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samplerate=self.gui_config.samplerate,
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dtype="float32",
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) as stream:
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global stream_latency
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stream_latency = stream.latency[-1]
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while self.flag_vc:
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time.sleep(self.gui_config.block_time)
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print("Audio block passed.")
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print("ENDing VC")
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def audio_callback(self, indata: np.ndarray, outdata: np.ndarray, frames, times, status):
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start_time = time.perf_counter()
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indata = librosa.to_mono(indata.T)
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if self.gui_config.threhold > -60:
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rms = librosa.feature.rms(y=indata, frame_length=4 * self.zc, hop_length=self.zc)
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db_threhold = (librosa.amplitude_to_db(rms, ref=1.0)[0] < self.gui_config.threhold)
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for i in range(db_threhold.shape[0]):
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if db_threhold[i]:
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indata[i * self.zc : (i + 1) * self.zc] = 0
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self.input_wav[: -self.block_frame] = self.input_wav[self.block_frame :].clone()
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self.input_wav[-self.block_frame :] = torch.from_numpy(indata).to(self.config.device)
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self.input_wav_res[: -self.block_frame_16k] = self.input_wav_res[self.block_frame_16k :].clone()
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if self.gui_config.I_noise_reduce and self.function == "vc":
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input_wav = self.input_wav[-self.crossfade_frame - self.block_frame - 2 * self.zc :]
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input_wav = self.tg(input_wav.unsqueeze(0), self.input_wav.unsqueeze(0))[0, 2 * self.zc :]
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input_wav[: self.crossfade_frame] *= self.fade_in_window
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input_wav[: self.crossfade_frame] += self.nr_buffer * self.fade_out_window
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self.nr_buffer[:] = input_wav[-self.crossfade_frame :]
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input_wav = torch.cat((self.res_buffer[:], input_wav[: self.block_frame]))
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self.res_buffer[:] = input_wav[-2 * self.zc :]
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self.input_wav_res[-self.block_frame_16k - 160 :] = self.resampler(input_wav)[160:]
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else:
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self.input_wav_res[-self.block_frame_16k - 160 :] = self.resampler(self.input_wav[-self.block_frame - 2 * self.zc :])[160:]
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if self.function == "vc":
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f0_extractor_frame = self.block_frame_16k + 800
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if self.gui_config.f0method == "rmvpe":
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f0_extractor_frame = (5120 * ((f0_extractor_frame - 1) // 5120 + 1) - 160)
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infer_wav = self.rvc.infer(
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self.input_wav_res,
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self.input_wav_res[-f0_extractor_frame:].cpu().numpy(),
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self.block_frame_16k,
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self.valid_rate,
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self.pitch,
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self.pitchf,
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self.gui_config.f0method,
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)
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infer_wav = infer_wav[-self.crossfade_frame - self.sola_search_frame - self.block_frame :]
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else:
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infer_wav = self.input_wav[-self.crossfade_frame - self.sola_search_frame - self.block_frame :].clone()
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if (self.gui_config.O_noise_reduce and self.function == "vc") or (self.gui_config.I_noise_reduce and self.function == "im"):
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self.output_buffer[: -self.block_frame] = self.output_buffer[self.block_frame :].clone()
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self.output_buffer[-self.block_frame :] = infer_wav[-self.block_frame :]
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infer_wav = self.tg(infer_wav.unsqueeze(0), self.output_buffer.unsqueeze(0)).squeeze(0)
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if self.gui_config.rms_mix_rate < 1 and self.function == "vc":
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rms1 = librosa.feature.rms(y=self.input_wav_res[-160 * infer_wav.shape[0] // self.zc :].cpu().numpy(), frame_length=640, hop_length=160)
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rms1 = torch.from_numpy(rms1).to(self.config.device)
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rms1 = F.interpolate(rms1.unsqueeze(0), size=infer_wav.shape[0] + 1, mode="linear", align_corners=True)[0, 0, :-1]
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rms2 = librosa.feature.rms(y=infer_wav[:].cpu().numpy(), frame_length=4 * self.zc, hop_length=self.zc)
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rms2 = torch.from_numpy(rms2).to(self.config.device)
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rms2 = F.interpolate(rms2.unsqueeze(0), size=infer_wav.shape[0] + 1, mode="linear", align_corners=True)[0, 0, :-1]
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rms2 = torch.max(rms2, torch.zeros_like(rms2) + 1e-3)
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infer_wav *= torch.pow(rms1 / rms2, torch.tensor(1 - self.gui_config.rms_mix_rate))
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conv_input = infer_wav[None, None, : self.crossfade_frame + self.sola_search_frame]
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cor_nom = F.conv1d(conv_input, self.sola_buffer[None, None, :])
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cor_den = torch.sqrt(F.conv1d(conv_input**2, torch.ones(1, 1, self.crossfade_frame, device=self.config.device)) + 1e-8)
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if sys.platform == "darwin":
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_, sola_offset = torch.max(cor_nom[0, 0] / cor_den[0, 0])
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sola_offset = sola_offset.item()
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else:
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sola_offset = torch.argmax(cor_nom[0, 0] / cor_den[0, 0])
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print("sola_offset = %d" % int(sola_offset))
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infer_wav = infer_wav[sola_offset : sola_offset + self.block_frame + self.crossfade_frame]
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infer_wav[: self.crossfade_frame] *= self.fade_in_window
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infer_wav[: self.crossfade_frame] += self.sola_buffer * self.fade_out_window
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self.sola_buffer[:] = infer_wav[-self.crossfade_frame :]
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if sys.platform == "darwin":
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outdata[:] = infer_wav[: -self.crossfade_frame].cpu().numpy()[:, np.newaxis]
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else:
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outdata[:] = infer_wav[: -self.crossfade_frame].repeat(2, 1).t().cpu().numpy()
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total_time = time.perf_counter() - start_time
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print("Infer time: %.2f" % total_time)
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def get_devices(self, update: bool = True):
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if update:
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sd._terminate()
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sd._initialize()
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devices = sd.query_devices()
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hostapis = sd.query_hostapis()
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for hostapi in hostapis:
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for device_idx in hostapi["devices"]:
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devices[device_idx]["hostapi_name"] = hostapi["name"]
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input_devices = [
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f"{d['name']} ({d['hostapi_name']})"
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for d in devices
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if d["max_input_channels"] > 0
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]
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output_devices = [
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f"{d['name']} ({d['hostapi_name']})"
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for d in devices
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if d["max_output_channels"] > 0
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]
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input_devices_indices = [
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d["index"] if "index" in d else d["name"]
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for d in devices
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if d["max_input_channels"] > 0
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]
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output_devices_indices = [
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d["index"] if "index" in d else d["name"]
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for d in devices
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if d["max_output_channels"] > 0
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]
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return (
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input_devices,
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output_devices,
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input_devices_indices,
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output_devices_indices,
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)
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def set_devices(self, input_device, output_device):
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(
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input_devices,
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output_devices,
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input_device_indices,
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output_device_indices,
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) = self.get_devices()
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sd.default.device[0] = input_device_indices[input_devices.index(input_device)]
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sd.default.device[1] = output_device_indices[output_devices.index(output_device)]
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print("Input device: %s:%s" % (str(sd.default.device[0]), input_device))
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print("Output device: %s:%s" % (str(sd.default.device[1]), output_device))
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audio_api = AudioAPI()
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@app.get("/inputDevices")
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def get_input_devices():
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input_devices, _, _, _ = audio_api.get_devices()
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return input_devices
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@app.get("/outputDevices")
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def get_output_devices():
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_, output_devices, _, _ = audio_api.get_devices()
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return output_devices
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@app.post("/config")
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def configure_audio(config_data: ConfigData):
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try:
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if audio_api.set_values(config_data):
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settings = config_data.dict()
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settings["use_jit"] = False
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settings["f0method"] = "rmvpe"
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with open("configs/config.json", "w") as j:
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json.dump(settings, j)
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return {"message": "Configuration set successfully"}
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except Exception as e:
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raise HTTPException(status_code=400, detail=f"Configuration failed: {e}")
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@app.post("/start")
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def start_conversion():
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try:
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if not audio_api.flag_vc:
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audio_api.start_vc()
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return {"message": "Audio conversion started"}
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else:
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raise HTTPException(status_code=400, detail="Audio conversion already running")
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except Exception as e:
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raise HTTPException(status_code=500, detail=f"Failed to start conversion: {e}")
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@app.post("/stop")
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def stop_conversion():
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try:
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if audio_api.flag_vc:
|
||||
audio_api.flag_vc = False
|
||||
global stream_latency
|
||||
stream_latency = -1
|
||||
return {"message": "Audio conversion stopped"}
|
||||
else:
|
||||
raise HTTPException(status_code=400, detail="Audio conversion not running")
|
||||
except Exception as e:
|
||||
raise HTTPException(status_code=500, detail=f"Failed to stop conversion: {e}")
|
||||
|
||||
if __name__ == "__main__":
|
||||
uvicorn.run(app, host="0.0.0.0", port=8043)
|
Loading…
Reference in New Issue
Block a user