from typing import Any, Union, cast import socketio from const import TMP_DIR, ModelType import torch import os import traceback import numpy as np from dataclasses import dataclass, asdict, field import resampy from voice_changer.IORecorder import IORecorder from voice_changer.Local.AudioDeviceList import ServerAudioDevice, list_audio_device from voice_changer.utils.LoadModelParams import LoadModelParams from voice_changer.utils.Timer import Timer from voice_changer.utils.VoiceChangerModel import VoiceChangerModel, AudioInOut from Exceptions import ( DeviceChangingException, HalfPrecisionChangingException, NoModeLoadedException, NotEnoughDataExtimateF0, ONNXInputArgumentException, ) from voice_changer.utils.VoiceChangerParams import VoiceChangerParams import threading import time import sounddevice as sd import librosa providers = [ "OpenVINOExecutionProvider", "CUDAExecutionProvider", "DmlExecutionProvider", "CPUExecutionProvider", ] STREAM_INPUT_FILE = os.path.join(TMP_DIR, "in.wav") STREAM_OUTPUT_FILE = os.path.join(TMP_DIR, "out.wav") @dataclass class VoiceChangerSettings: inputSampleRate: int = 48000 # 48000 or 24000 crossFadeOffsetRate: float = 0.1 crossFadeEndRate: float = 0.9 crossFadeOverlapSize: int = 4096 recordIO: int = 0 # 0:off, 1:on serverAudioInputDevices: list[ServerAudioDevice] = field(default_factory=lambda: []) serverAudioOutputDevices: list[ServerAudioDevice] = field( default_factory=lambda: [] ) enableServerAudio: int = 0 # 0:off, 1:on serverAudioStated: int = 0 # 0:off, 1:on # serverInputAudioSampleRate: int = 48000 # serverOutputAudioSampleRate: int = 48000 serverInputAudioSampleRate: int = 44100 serverOutputAudioSampleRate: int = 44100 serverInputAudioBufferSize: int = 1024 * 24 serverOutputAudioBufferSize: int = 1024 * 24 serverInputDeviceId: int = -1 serverOutputDeviceId: int = -1 serverReadChunkSize: int = 256 performance: list[int] = field(default_factory=lambda: [0, 0, 0, 0]) # ↓mutableな物だけ列挙 intData: list[str] = field( default_factory=lambda: [ "inputSampleRate", "crossFadeOverlapSize", "recordIO", "enableServerAudio", "serverAudioStated", "serverInputAudioSampleRate", "serverOutputAudioSampleRate", "serverInputAudioBufferSize", "serverOutputAudioBufferSize", "serverInputDeviceId", "serverOutputDeviceId", "serverReadChunkSize", ] ) floatData: list[str] = field( default_factory=lambda: ["crossFadeOffsetRate", "crossFadeEndRate"] ) strData: list[str] = field(default_factory=lambda: []) class VoiceChanger: settings: VoiceChangerSettings voiceChanger: VoiceChangerModel ioRecorder: IORecorder sola_buffer: AudioInOut namespace: socketio.AsyncNamespace | None = None def audio_callback( self, indata: np.ndarray, outdata: np.ndarray, frames, times, status ): # print(indata) try: with Timer("all_inference_time") as t: unpackedData = librosa.to_mono(indata.T) * 32768.0 out_wav, times = self.on_request(unpackedData) outdata[:] = np.repeat(out_wav, 2).reshape(-1, 2) / 32768.0 all_inference_time = t.secs performance = [all_inference_time] + times performance = [round(x * 1000) for x in performance] except Exception as e: print(e) def serverLocal(self, _vc): vc: VoiceChanger = _vc currentInputDeviceId = -1 currentInputSampleRate = -1 currentOutputDeviceId = -1 currentInputChunkNum = -1 while True: if ( vc.settings.serverAudioStated == 0 or vc.settings.serverInputDeviceId == -1 ): vc.settings.inputSampleRate = 48000 time.sleep(2) else: sd._terminate() sd._initialize() if currentInputDeviceId != vc.settings.serverInputDeviceId: sd.default.device[0] = vc.settings.serverInputDeviceId currentInputDeviceId = vc.settings.serverInputDeviceId if currentOutputDeviceId != vc.settings.serverOutputDeviceId: sd.default.device[1] = vc.settings.serverOutputDeviceId currentOutputDeviceId = vc.settings.serverOutputDeviceId currentInputSampleRate = vc.settings.serverInputAudioSampleRate currentInputChunkNum = vc.settings.serverReadChunkSize block_frame = currentInputChunkNum * 128 try: with sd.Stream( callback=self.audio_callback, blocksize=block_frame, samplerate=currentInputSampleRate, dtype="float32", ): while ( vc.settings.serverAudioStated == 1 and currentInputDeviceId == vc.settings.serverInputDeviceId and currentOutputDeviceId == vc.settings.serverOutputDeviceId and currentInputSampleRate == vc.settings.serverInputAudioSampleRate and currentInputChunkNum == vc.settings.serverReadChunkSize ): vc.settings.serverInputAudioSampleRate = ( self.voiceChanger.get_processing_sampling_rate() ) vc.settings.inputSampleRate = ( vc.settings.serverInputAudioSampleRate ) time.sleep(2) print( "[Voice Changer] server audio", self.settings.performance, ) print( "[Voice Changer] info:", vc.settings.serverAudioStated, currentInputDeviceId, currentOutputDeviceId, currentInputSampleRate, currentInputChunkNum, ) except Exception as e: print(e) time.sleep(2) def __init__(self, params: VoiceChangerParams): # 初期化 self.settings = VoiceChangerSettings() self.onnx_session = None self.currentCrossFadeOffsetRate = 0.0 self.currentCrossFadeEndRate = 0.0 self.currentCrossFadeOverlapSize = 0 # setting self.crossfadeSize = 0 # calculated self.voiceChanger = None self.modelType: ModelType | None = None self.params = params self.gpu_num = torch.cuda.device_count() self.prev_audio = np.zeros(4096) self.mps_enabled: bool = ( getattr(torch.backends, "mps", None) is not None and torch.backends.mps.is_available() ) audioinput, audiooutput = list_audio_device() self.settings.serverAudioInputDevices = audioinput self.settings.serverAudioOutputDevices = audiooutput thread = threading.Thread(target=self.serverLocal, args=(self,)) thread.start() print( f"VoiceChanger Initialized (GPU_NUM:{self.gpu_num}, mps_enabled:{self.mps_enabled})" ) def switchModelType(self, modelType: ModelType): try: print("switch model type", 1) if hasattr(self, "voiceChanger") and self.voiceChanger is not None: # return {"status": "ERROR", "msg": "vc is already selected. currently re-select is not implemented"} del self.voiceChanger self.voiceChanger = None self.modelType = modelType if self.modelType == "MMVCv15": from voice_changer.MMVCv15.MMVCv15 import MMVCv15 self.voiceChanger = MMVCv15() # type: ignore elif self.modelType == "MMVCv13": from voice_changer.MMVCv13.MMVCv13 import MMVCv13 self.voiceChanger = MMVCv13() elif self.modelType == "so-vits-svc-40v2": from voice_changer.SoVitsSvc40v2.SoVitsSvc40v2 import SoVitsSvc40v2 self.voiceChanger = SoVitsSvc40v2(self.params) elif ( self.modelType == "so-vits-svc-40" or self.modelType == "so-vits-svc-40_c" ): from voice_changer.SoVitsSvc40.SoVitsSvc40 import SoVitsSvc40 self.voiceChanger = SoVitsSvc40(self.params) elif self.modelType == "DDSP-SVC": print("switch model type", 2) from voice_changer.DDSP_SVC.DDSP_SVC import DDSP_SVC self.voiceChanger = DDSP_SVC(self.params) elif self.modelType == "RVC": from voice_changer.RVC.RVC import RVC self.voiceChanger = RVC(self.params) else: from voice_changer.MMVCv13.MMVCv13 import MMVCv13 self.voiceChanger = MMVCv13() except Exception as e: print(e) print(traceback.format_exc()) return {"status": "OK", "msg": "vc is switched."} def getModelType(self): if self.modelType is not None: return {"status": "OK", "vc": self.modelType} else: return {"status": "OK", "vc": "none"} def loadModel(self, props: LoadModelParams): try: return self.voiceChanger.loadModel(props) except Exception as e: print(traceback.format_exc()) print("[Voice Changer] Model Load Error! Check your model is valid.", e) return {"status": "NG"} def get_info(self): data = asdict(self.settings) if hasattr(self, "voiceChanger"): data.update(self.voiceChanger.get_info()) return data def get_performance(self): return self.settings.performance def update_settings(self, key: str, val: Any): if key in self.settings.intData: setattr(self.settings, key, int(val)) if key == "crossFadeOffsetRate" or key == "crossFadeEndRate": self.crossfadeSize = 0 if key == "recordIO" and val == 1: if hasattr(self, "ioRecorder"): self.ioRecorder.close() self.ioRecorder = IORecorder( STREAM_INPUT_FILE, STREAM_OUTPUT_FILE, self.settings.inputSampleRate ) if key == "recordIO" and val == 0: if hasattr(self, "ioRecorder"): self.ioRecorder.close() pass if key == "recordIO" and val == 2: if hasattr(self, "ioRecorder"): self.ioRecorder.close() elif key in self.settings.floatData: setattr(self.settings, key, float(val)) elif key in self.settings.strData: setattr(self.settings, key, str(val)) else: if hasattr(self, "voiceChanger"): ret = self.voiceChanger.update_settings(key, val) if ret is False: print(f"{key} is not mutable variable or unknown variable!") else: print("voice changer is not initialized!") return self.get_info() def _generate_strength(self, crossfadeSize: int): if ( self.crossfadeSize != crossfadeSize or self.currentCrossFadeOffsetRate != self.settings.crossFadeOffsetRate or self.currentCrossFadeEndRate != self.settings.crossFadeEndRate or self.currentCrossFadeOverlapSize != self.settings.crossFadeOverlapSize ): self.crossfadeSize = crossfadeSize self.currentCrossFadeOffsetRate = self.settings.crossFadeOffsetRate self.currentCrossFadeEndRate = self.settings.crossFadeEndRate self.currentCrossFadeOverlapSize = self.settings.crossFadeOverlapSize cf_offset = int(crossfadeSize * self.settings.crossFadeOffsetRate) cf_end = int(crossfadeSize * self.settings.crossFadeEndRate) cf_range = cf_end - cf_offset percent = np.arange(cf_range) / cf_range np_prev_strength = np.cos(percent * 0.5 * np.pi) ** 2 np_cur_strength = np.cos((1 - percent) * 0.5 * np.pi) ** 2 self.np_prev_strength = np.concatenate( [ np.ones(cf_offset), np_prev_strength, np.zeros(crossfadeSize - cf_offset - len(np_prev_strength)), ] ) self.np_cur_strength = np.concatenate( [ np.zeros(cf_offset), np_cur_strength, np.ones(crossfadeSize - cf_offset - len(np_cur_strength)), ] ) print( f"Generated Strengths: for prev:{self.np_prev_strength.shape}, for cur:{self.np_cur_strength.shape}" ) # ひとつ前の結果とサイズが変わるため、記録は消去する。 if hasattr(self, "np_prev_audio1") is True: delattr(self, "np_prev_audio1") if hasattr(self, "sola_buffer") is True: del self.sola_buffer # receivedData: tuple of short def on_request( self, receivedData: AudioInOut ) -> tuple[AudioInOut, list[Union[int, float]]]: return self.on_request_sola(receivedData) def on_request_sola( self, receivedData: AudioInOut ) -> tuple[AudioInOut, list[Union[int, float]]]: try: processing_sampling_rate = self.voiceChanger.get_processing_sampling_rate() # 前処理 with Timer("pre-process") as t: if self.settings.inputSampleRate != processing_sampling_rate: newData = cast( AudioInOut, resampy.resample( receivedData, self.settings.inputSampleRate, processing_sampling_rate, ), ) else: newData = receivedData sola_search_frame = int(0.012 * processing_sampling_rate) # sola_search_frame = 0 block_frame = newData.shape[0] crossfade_frame = min(self.settings.crossFadeOverlapSize, block_frame) self._generate_strength(crossfade_frame) data = self.voiceChanger.generate_input( newData, block_frame, crossfade_frame, sola_search_frame ) preprocess_time = t.secs # 変換処理 with Timer("main-process") as t: # Inference audio = self.voiceChanger.inference(data) if hasattr(self, "sola_buffer") is True: np.set_printoptions(threshold=10000) audio_offset = -1 * ( sola_search_frame + crossfade_frame + block_frame ) audio = audio[audio_offset:] a = 0 audio = audio[a:] # SOLA algorithm from https://github.com/yxlllc/DDSP-SVC, https://github.com/liujing04/Retrieval-based-Voice-Conversion-WebUI cor_nom = np.convolve( audio[: crossfade_frame + sola_search_frame], np.flip(self.sola_buffer), "valid", ) cor_den = np.sqrt( np.convolve( audio[: crossfade_frame + sola_search_frame] ** 2, np.ones(crossfade_frame), "valid", ) + 1e-3 ) sola_offset = int(np.argmax(cor_nom / cor_den)) sola_end = sola_offset + block_frame output_wav = audio[sola_offset:sola_end].astype(np.float64) output_wav[:crossfade_frame] *= self.np_cur_strength output_wav[:crossfade_frame] += self.sola_buffer[:] result = output_wav else: print("[Voice Changer] no sola buffer. (You can ignore this.)") result = np.zeros(4096).astype(np.int16) if ( hasattr(self, "sola_buffer") is True and sola_offset < sola_search_frame ): offset = -1 * (sola_search_frame + crossfade_frame - sola_offset) end = -1 * (sola_search_frame - sola_offset) sola_buf_org = audio[offset:end] self.sola_buffer = sola_buf_org * self.np_prev_strength else: self.sola_buffer = audio[-crossfade_frame:] * self.np_prev_strength # self.sola_buffer = audio[- crossfade_frame:] mainprocess_time = t.secs # 後処理 with Timer("post-process") as t: result = result.astype(np.int16) if self.settings.inputSampleRate != processing_sampling_rate: # print( # "samplingrate", # self.settings.inputSampleRate, # processing_sampling_rate, # ) outputData = cast( AudioInOut, resampy.resample( result, processing_sampling_rate, self.settings.inputSampleRate, ).astype(np.int16), ) else: outputData = result print_convert_processing( f" Output data size of {result.shape[0]}/{processing_sampling_rate}hz {outputData.shape[0]}/{self.settings.inputSampleRate}hz" ) if self.settings.recordIO == 1: self.ioRecorder.writeInput(receivedData) self.ioRecorder.writeOutput(outputData.tobytes()) if receivedData.shape[0] != outputData.shape[0]: # print( # f"Padding, in:{receivedData.shape[0]} out:{outputData.shape[0]}" # ) outputData = pad_array(outputData, receivedData.shape[0]) # print_convert_processing( # f" Padded!, Output data size of {result.shape[0]}/{processing_sampling_rate}hz {outputData.shape[0]}/{self.settings.inputSampleRate}hz") postprocess_time = t.secs print_convert_processing( f" [fin] Input/Output size:{receivedData.shape[0]},{outputData.shape[0]}" ) perf = [preprocess_time, mainprocess_time, postprocess_time] return outputData, perf except NoModeLoadedException as e: print("[Voice Changer] [Exception]", e) return np.zeros(1).astype(np.int16), [0, 0, 0] except ONNXInputArgumentException as e: print("[Voice Changer] [Exception]", e) return np.zeros(1).astype(np.int16), [0, 0, 0] except HalfPrecisionChangingException as e: print("[Voice Changer] Switching model configuration....", e) return np.zeros(1).astype(np.int16), [0, 0, 0] except NotEnoughDataExtimateF0 as e: print("[Voice Changer] not enough data", e) return np.zeros(1).astype(np.int16), [0, 0, 0] except DeviceChangingException as e: print("[Voice Changer] embedder:", e) return np.zeros(1).astype(np.int16), [0, 0, 0] except Exception as e: print("VC PROCESSING!!!! EXCEPTION!!!", e) print(traceback.format_exc()) return np.zeros(1).astype(np.int16), [0, 0, 0] def export2onnx(self): return self.voiceChanger.export2onnx() ############## def merge_models(self, request: str): self.voiceChanger.merge_models(request) return self.get_info() PRINT_CONVERT_PROCESSING: bool = False # PRINT_CONVERT_PROCESSING = True def print_convert_processing(mess: str): if PRINT_CONVERT_PROCESSING is True: print(mess) def pad_array(arr: AudioInOut, target_length: int): current_length = arr.shape[0] if current_length >= target_length: return arr else: pad_width = target_length - current_length pad_left = pad_width // 2 pad_right = pad_width - pad_left padded_arr = np.pad( arr, (pad_left, pad_right), "constant", constant_values=(0, 0) ) return padded_arr