| import os | |
| import sys | |
| import time | |
| import torch | |
| import logging | |
| import numpy as np | |
| import soundfile as sf | |
| import librosa | |
| now_dir = os.getcwd() | |
| sys.path.append(now_dir) | |
| from rvc.infer.pipeline import VC | |
| from scipy.io import wavfile | |
| from audio_upscaler import upscale | |
| import noisereduce as nr | |
| from rvc.lib.utils import load_audio | |
| from rvc.lib.tools.split_audio import process_audio, merge_audio | |
| from rvc.lib.infer_pack.models import ( | |
| SynthesizerTrnMs256NSFsid, | |
| SynthesizerTrnMs256NSFsid_nono, | |
| SynthesizerTrnMs768NSFsid, | |
| SynthesizerTrnMs768NSFsid_nono, | |
| ) | |
| from rvc.configs.config import Config | |
| from rvc.lib.utils import load_embedding | |
| logging.getLogger("httpx").setLevel(logging.WARNING) | |
| logging.getLogger("httpcore").setLevel(logging.WARNING) | |
| config = Config() | |
| hubert_model = None | |
| tgt_sr = None | |
| net_g = None | |
| vc = None | |
| cpt = None | |
| version = None | |
| n_spk = None | |
| def load_hubert(embedder_model, embedder_model_custom): | |
| global hubert_model | |
| models, _, _ = load_embedding(embedder_model, embedder_model_custom) | |
| 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() | |
| hubert_model.eval() | |
| def remove_audio_noise(input_audio_path, reduction_strength=0.7): | |
| try: | |
| rate, data = wavfile.read(input_audio_path) | |
| reduced_noise = nr.reduce_noise( | |
| y=data, | |
| sr=rate, | |
| prop_decrease=reduction_strength, | |
| ) | |
| return reduced_noise | |
| except Exception as error: | |
| print(f"Error cleaning audio: {error}") | |
| return None | |
| def convert_audio_format(input_path, output_path, output_format): | |
| try: | |
| if output_format != "WAV": | |
| print(f"Converting audio to {output_format} format...") | |
| audio, sample_rate = librosa.load(input_path, sr=None) | |
| common_sample_rates = [ | |
| 8000, | |
| 11025, | |
| 12000, | |
| 16000, | |
| 22050, | |
| 24000, | |
| 32000, | |
| 44100, | |
| 48000, | |
| ] | |
| target_sr = min(common_sample_rates, key=lambda x: abs(x - sample_rate)) | |
| audio = librosa.resample(audio, orig_sr=sample_rate, target_sr=target_sr) | |
| sf.write(output_path, audio, target_sr, format=output_format.lower()) | |
| return output_path | |
| except Exception as error: | |
| print(f"Failed to convert audio to {output_format} format: {error}") | |
| def voice_conversion( | |
| sid=0, | |
| input_audio_path=None, | |
| f0_up_key=None, | |
| f0_file=None, | |
| f0_method=None, | |
| file_index=None, | |
| index_rate=None, | |
| resample_sr=0, | |
| rms_mix_rate=None, | |
| protect=None, | |
| hop_length=None, | |
| output_path=None, | |
| split_audio=False, | |
| f0autotune=False, | |
| filter_radius=None, | |
| embedder_model=None, | |
| embedder_model_custom=None, | |
| ): | |
| global tgt_sr, net_g, vc, hubert_model, version | |
| f0_up_key = int(f0_up_key) | |
| try: | |
| audio = load_audio(input_audio_path, 16000) | |
| audio_max = np.abs(audio).max() / 0.95 | |
| if audio_max > 1: | |
| audio /= audio_max | |
| if not hubert_model: | |
| load_hubert(embedder_model, embedder_model_custom) | |
| if_f0 = cpt.get("f0", 1) | |
| file_index = ( | |
| file_index.strip(" ") | |
| .strip('"') | |
| .strip("\n") | |
| .strip('"') | |
| .strip(" ") | |
| .replace("trained", "added") | |
| ) | |
| if tgt_sr != resample_sr >= 16000: | |
| tgt_sr = resample_sr | |
| if split_audio == "True": | |
| result, new_dir_path = process_audio(input_audio_path) | |
| if result == "Error": | |
| return "Error with Split Audio", None | |
| dir_path = ( | |
| new_dir_path.strip(" ").strip('"').strip("\n").strip('"').strip(" ") | |
| ) | |
| if dir_path != "": | |
| paths = [ | |
| os.path.join(root, name) | |
| for root, _, files in os.walk(dir_path, topdown=False) | |
| for name in files | |
| if name.endswith(".wav") and root == dir_path | |
| ] | |
| try: | |
| for path in paths: | |
| voice_conversion( | |
| sid, | |
| path, | |
| f0_up_key, | |
| None, | |
| f0_method, | |
| file_index, | |
| index_rate, | |
| resample_sr, | |
| rms_mix_rate, | |
| protect, | |
| hop_length, | |
| path, | |
| False, | |
| f0autotune, | |
| filter_radius, | |
| embedder_model, | |
| embedder_model_custom, | |
| ) | |
| except Exception as error: | |
| print(error) | |
| return f"Error {error}" | |
| print("Finished processing segmented audio, now merging audio...") | |
| merge_timestamps_file = os.path.join( | |
| os.path.dirname(new_dir_path), | |
| f"{os.path.basename(input_audio_path).split('.')[0]}_timestamps.txt", | |
| ) | |
| tgt_sr, audio_opt = merge_audio(merge_timestamps_file) | |
| os.remove(merge_timestamps_file) | |
| else: | |
| audio_opt = vc.pipeline( | |
| hubert_model, | |
| net_g, | |
| sid, | |
| audio, | |
| input_audio_path, | |
| f0_up_key, | |
| f0_method, | |
| file_index, | |
| index_rate, | |
| if_f0, | |
| filter_radius, | |
| tgt_sr, | |
| resample_sr, | |
| rms_mix_rate, | |
| version, | |
| protect, | |
| hop_length, | |
| f0autotune, | |
| f0_file=f0_file, | |
| ) | |
| if output_path is not None: | |
| sf.write(output_path, audio_opt, tgt_sr, format="WAV") | |
| return (tgt_sr, audio_opt) | |
| except Exception as error: | |
| print(error) | |
| def get_vc(weight_root, sid): | |
| global n_spk, tgt_sr, net_g, vc, cpt, version | |
| if sid == "" or sid == []: | |
| global hubert_model | |
| if hubert_model is not None: | |
| print("clean_empty_cache") | |
| del net_g, n_spk, vc, hubert_model, tgt_sr | |
| hubert_model = net_g = n_spk = vc = hubert_model = tgt_sr = None | |
| if torch.cuda.is_available(): | |
| torch.cuda.empty_cache() | |
| 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"]) | |
| del net_g, cpt | |
| if torch.cuda.is_available(): | |
| torch.cuda.empty_cache() | |
| cpt = None | |
| person = weight_root | |
| cpt = torch.load(person, map_location="cpu") | |
| tgt_sr = cpt["config"][-1] | |
| cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] | |
| 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"]) | |
| del net_g.enc_q | |
| print(net_g.load_state_dict(cpt["weight"], strict=False)) | |
| 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) | |
| n_spk = cpt["config"][-3] | |
| def infer_pipeline( | |
| f0up_key, | |
| filter_radius, | |
| index_rate, | |
| rms_mix_rate, | |
| protect, | |
| hop_length, | |
| f0method, | |
| audio_input_path, | |
| audio_output_path, | |
| model_path, | |
| index_path, | |
| split_audio, | |
| f0autotune, | |
| clean_audio, | |
| clean_strength, | |
| export_format, | |
| embedder_model, | |
| embedder_model_custom, | |
| upscale_audio, | |
| ): | |
| global tgt_sr, net_g, vc, cpt | |
| get_vc(model_path, 0) | |
| try: | |
| if upscale_audio == "True": | |
| upscale(audio_input_path, audio_input_path) | |
| start_time = time.time() | |
| voice_conversion( | |
| sid=0, | |
| input_audio_path=audio_input_path, | |
| f0_up_key=f0up_key, | |
| f0_file=None, | |
| f0_method=f0method, | |
| file_index=index_path, | |
| index_rate=float(index_rate), | |
| rms_mix_rate=float(rms_mix_rate), | |
| protect=float(protect), | |
| hop_length=hop_length, | |
| output_path=audio_output_path, | |
| split_audio=split_audio, | |
| f0autotune=f0autotune, | |
| filter_radius=filter_radius, | |
| embedder_model=embedder_model, | |
| embedder_model_custom=embedder_model_custom, | |
| ) | |
| if clean_audio == "True": | |
| cleaned_audio = remove_audio_noise(audio_output_path, clean_strength) | |
| if cleaned_audio is not None: | |
| sf.write(audio_output_path, cleaned_audio, tgt_sr, format="WAV") | |
| output_path_format = audio_output_path.replace( | |
| ".wav", f".{export_format.lower()}" | |
| ) | |
| audio_output_path = convert_audio_format( | |
| audio_output_path, output_path_format, export_format | |
| ) | |
| end_time = time.time() | |
| elapsed_time = end_time - start_time | |
| print( | |
| f"Conversion completed. Output file: '{audio_output_path}' in {elapsed_time:.2f} seconds." | |
| ) | |
| except Exception as error: | |
| print(f"Voice conversion failed: {error}") | |