initial upload attempt
initial upload attempt
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erew123 committed
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import os
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import sys
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import time
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import torch
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import logging
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import faiss
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import numpy as np
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import soundfile as sf
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import librosa
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now_dir = os.getcwd()
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sys.path.append(now_dir)
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from ..infer.pipeline import VC
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from ..lib.utils import load_audio
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from ..lib.tools.split_audio import process_audio, merge_audio
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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 ..configs.config import Config
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from ..lib.utils import load_embedding
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logging.getLogger("httpx").setLevel(logging.WARNING)
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logging.getLogger("httpcore").setLevel(logging.WARNING)
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config = Config()
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hubert_model = None
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tgt_sr = None
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net_g = None
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vc = None
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cpt = None
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version = None
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n_spk = None
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debug_rvc = False
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def load_hubert(embedder_model):
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global hubert_model
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models, _, _ = load_embedding(embedder_model)
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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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hubert_model.eval()
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def voice_conversion(
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sid=0,
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input_audio_path=None,
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f0_up_key=None,
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f0_file=None,
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f0_method=None,
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file_index=None,
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index_rate=None,
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resample_sr=0,
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rms_mix_rate=None,
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protect=None,
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hop_length=None,
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output_path=None,
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split_audio=False,
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f0autotune=False,
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filter_radius=None,
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embedder_model=None,
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):
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global tgt_sr, net_g, vc, hubert_model, version
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f0_up_key = int(f0_up_key)
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try:
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print(f"Loading audio from {input_audio_path}") if debug_rvc else None
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audio = load_audio(input_audio_path, 16000)
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audio_max = np.abs(audio).max() / 0.95
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if audio_max > 1:
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audio /= audio_max
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if not hubert_model:
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print(f"Loading hubert model with {embedder_model}") if debug_rvc else None
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load_hubert(embedder_model)
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if_f0 = cpt.get("f0", 1)
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file_index = (
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file_index.strip(" ")
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.strip('"')
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.strip("\n")
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.strip('"')
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.strip(" ")
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.replace("trained", "added")
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)
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if tgt_sr != resample_sr >= 16000:
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tgt_sr = resample_sr
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if split_audio == "True":
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print("Splitting audio") if debug_rvc else None
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result, new_dir_path = process_audio(input_audio_path)
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if result == "Error":
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return "Error with Split Audio", None
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dir_path = (
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new_dir_path.strip(" ").strip('"').strip("\n").strip('"').strip(" ")
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)
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if dir_path != "":
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paths = [
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os.path.join(root, name)
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for root, _, files in os.walk(dir_path, topdown=False)
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for name in files
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if name.endswith(".wav") and root == dir_path
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]
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try:
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for path in paths:
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voice_conversion(
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sid,
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path,
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f0_up_key,
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None,
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f0_method,
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file_index,
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index_rate,
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resample_sr,
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rms_mix_rate,
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protect,
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hop_length,
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path,
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False,
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f0autotune,
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embedder_model,
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)
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except Exception as error:
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print(f"Error processing segmented audio: {error}")
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return f"Error {error}"
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print("Finished processing segmented audio, now merging audio...") if debug_rvc else None
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merge_timestamps_file = os.path.join(
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os.path.dirname(new_dir_path),
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f"{os.path.basename(input_audio_path).split('.')[0]}_timestamps.txt",
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)
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tgt_sr, audio_opt = merge_audio(merge_timestamps_file)
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os.remove(merge_timestamps_file)
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else:
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print("Processing audio with VC pipeline") if debug_rvc else None
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audio_opt = vc.pipeline(
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hubert_model,
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net_g,
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sid,
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audio,
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input_audio_path,
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f0_up_key,
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f0_method,
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file_index,
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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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hop_length,
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f0autotune,
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f0_file=f0_file,
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)
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# Resample the audio to the target sample rate before saving
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if tgt_sr != resample_sr and resample_sr >= 16000:
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print(f"Resampling audio from {tgt_sr} to {resample_sr}") if debug_rvc else None
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audio_opt = librosa.resample(audio_opt, tgt_sr, resample_sr)
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tgt_sr = resample_sr
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if output_path is not None:
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print(f"Saving file to {output_path}") if debug_rvc else None
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sf.write(output_path, audio_opt, tgt_sr, format="WAV")
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print(f"File saved to {output_path}") if debug_rvc else None
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return (tgt_sr, audio_opt)
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except Exception as error:
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print(f"Error during voice conversion: {error}")
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return None, None
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def get_vc(weight_root, sid, file_index=None, training_data_size=10000, debug_rvc=False):
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global n_spk, tgt_sr, net_g, vc, cpt, version
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branding="AllTalk "
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if debug_rvc:
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print(f"[{branding}Debug] Entering get_vc function")
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print(f"[{branding}Debug] weight_root: {weight_root}")
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print(f"[{branding}Debug] sid: {sid}")
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print(f"[{branding}Debug] file_index: {file_index}")
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print(f"[{branding}Debug] training_data_size: {training_data_size}")
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if sid == "" or sid == []:
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global hubert_model
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if hubert_model is not None:
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if debug_rvc:
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print(f"[{branding}Debug] Cleaning empty cache")
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del net_g, n_spk, vc, hubert_model, tgt_sr
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hubert_model = net_g = n_spk = vc = hubert_model = tgt_sr = None
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if torch.cuda.is_available():
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torch.cuda.empty_cache()
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if debug_rvc:
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print(f"[{branding}Debug] sid is empty or an empty list")
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if_f0 = cpt.get("f0", 1)
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version = cpt.get("version", "v1")
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if debug_rvc:
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print(f"[{branding}Debug] Model version: {version}")
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print(f"[{branding}Debug] f0: {if_f0}")
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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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del net_g, cpt
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if torch.cuda.is_available():
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torch.cuda.empty_cache()
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cpt = None
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if debug_rvc:
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print(f"[{branding}Debug] Loading model checkpoint")
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person = weight_root
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cpt = torch.load(person, 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 debug_rvc:
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print(f"[{branding}Debug] Model version: {version}")
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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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net_g.load_state_dict(cpt["weight"], strict=False)
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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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if file_index is not None:
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if debug_rvc:
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print(f"[{branding}Debug] Loading index file: {file_index}")
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index_file = file_index
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index = faiss.read_index(index_file)
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if debug_rvc:
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print(f"[{branding}Debug] Extracting data from index")
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data = index.reconstruct_n(0, index.ntotal)
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d = index.d
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nlist = index.nlist
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training_data_size = min(training_data_size, len(data))
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train_data = data[:training_data_size]
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else:
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if debug_rvc:
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print(f"[{branding}Debug] No index file provided")
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data = None
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d = None
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nlist = None
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train_data = None
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if debug_rvc:
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print(f"[{branding}Debug] Creating VC instance")
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if data is not None and d is not None and nlist is not None and train_data is not None:
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if debug_rvc:
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print(f"[{branding}Debug] Data: {data}")
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print(f"[{branding}Debug] Dimensionality: {d}")
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print(f"[{branding}Debug] Number of centroids: {nlist}")
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print(f"[{branding}Debug] Training data size: {len(train_data)}")
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vc = VC(tgt_sr, config, data, d, train_data, debug_rvc)
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else:
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vc = VC(tgt_sr, config)
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n_spk = cpt["config"][-3]
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if debug_rvc:
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print(f"[{branding}Debug] Leaving get_vc function")
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return vc
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def infer_pipeline(
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f0up_key,
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filter_radius,
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index_rate,
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rms_mix_rate,
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protect,
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hop_length,
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f0method,
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audio_input_path,
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audio_output_path,
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model_path,
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index_path,
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split_audio,
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f0autotune,
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embedder_model,
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training_data_size,
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debug_rvc,
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):
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global tgt_sr, net_g, vc, cpt
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debug_rvc = debug_rvc
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if index_path is None or index_path.strip() == "":
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file_index = None
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else:
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file_index = index_path
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get_vc(model_path, 0, file_index, training_data_size, debug_rvc)
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try:
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start_time = time.time()
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voice_conversion(
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sid=0,
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input_audio_path=audio_input_path,
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f0_up_key=f0up_key,
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f0_file=None,
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f0_method=f0method,
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file_index=index_path,
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index_rate=float(index_rate),
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rms_mix_rate=float(rms_mix_rate),
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protect=float(protect),
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hop_length=hop_length,
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output_path=audio_output_path,
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split_audio=split_audio,
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f0autotune=f0autotune,
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filter_radius=filter_radius,
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embedder_model=embedder_model,
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)
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end_time = time.time()
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elapsed_time = end_time - start_time
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print(f"Conversion completed. Output file: '{audio_output_path}' in {elapsed_time:.2f} seconds.") if debug_rvc else None
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except Exception as error:
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print(f"Voice conversion failed: {error}")
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