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import os
import sys
import importlib.util
import site
import json
import torch
import gradio as gr
import torchaudio
import numpy as np
from huggingface_hub import snapshot_download, hf_hub_download
import subprocess
import re
import spaces
import uuid
import soundfile as sf
# اضافه شدن کتابخانه PyDub
from pydub import AudioSegment
import io
# --- نصب و پچ کردن ---
downloaded_resources = {
"configs": False,
"tokenizer_vq8192": False,
"fmt_Vq8192ToMels": False,
"vocoder": False
}
def install_espeak():
try:
result = subprocess.run(["which", "espeak-ng"], capture_output=True, text=True)
if result.returncode != 0:
print("Installing espeak-ng...")
subprocess.run(["apt-get", "update"], check=True)
subprocess.run(["apt-get", "install", "-y", "espeak-ng", "espeak-ng-data"], check=True)
except Exception as e:
print(f"Error installing espeak-ng: {e}")
install_espeak()
def patch_langsegment_init():
try:
spec = importlib.util.find_spec("LangSegment")
if spec is None or spec.origin is None: return
init_path = os.path.join(os.path.dirname(spec.origin), '__init__.py')
with open(init_path, 'r') as f: lines = f.readlines()
modified = False
new_lines = []
target_line_prefix = "from .LangSegment import"
for line in lines:
if line.strip().startswith(target_line_prefix) and ('setLangfilters' in line or 'getLangfilters' in line):
mod_line = line.replace(',setLangfilters', '').replace(',getLangfilters', '')
mod_line = mod_line.replace('setLangfilters,', '').replace('getLangfilters,', '').rstrip(',')
new_lines.append(mod_line + '\n')
modified = True
else:
new_lines.append(line)
if modified:
with open(init_path, 'w') as f: f.writelines(new_lines)
try:
import LangSegment
importlib.reload(LangSegment)
except: pass
except: pass
patch_langsegment_init()
if not os.path.exists("Amphion"):
subprocess.run(["git", "clone", "https://github.com/open-mmlab/Amphion.git"])
os.chdir("Amphion")
if os.path.dirname(os.path.abspath("Amphion")) not in sys.path:
sys.path.append(os.path.dirname(os.path.abspath("Amphion")))
os.makedirs("wav", exist_ok=True)
os.makedirs("ckpts/Vevo", exist_ok=True)
from models.vc.vevo.vevo_utils import VevoInferencePipeline
# --- توابع کمکی جدید برای PyDub ---
def numpy_to_audiosegment(audio_arr, sample_rate=24000):
"""تبدیل آرایه نامپای (Float32) به آبجکت AudioSegment"""
# تبدیل به PCM 16-bit
audio_int16 = (audio_arr * 32767).astype(np.int16)
# ایجاد فایل در حافظه
byte_io = io.BytesIO()
sf.write(byte_io, audio_int16, sample_rate, format='WAV', subtype='PCM_16')
byte_io.seek(0)
return AudioSegment.from_wav(byte_io)
def save_audio_pcm16(waveform, output_path, sample_rate=24000):
# این تابع فقط برای ذخیره فایلهای موقت ورودی مدل است
try:
if isinstance(waveform, torch.Tensor):
waveform = waveform.detach().cpu().squeeze().numpy()
sf.write(output_path, waveform, sample_rate, subtype='PCM_16')
except Exception as e:
print(f"Save error: {e}")
def setup_configs():
if downloaded_resources["configs"]: return
config_path = "models/vc/vevo/config"
os.makedirs(config_path, exist_ok=True)
config_files = ["Vq8192ToMels.json", "Vocoder.json"]
for file in config_files:
file_path = f"{config_path}/{file}"
if not os.path.exists(file_path):
try:
file_data = hf_hub_download(repo_id="amphion/Vevo", filename=f"config/{file}", repo_type="model")
subprocess.run(["cp", file_data, file_path])
except Exception as e: print(f"Error downloading config {file}: {e}")
downloaded_resources["configs"] = True
setup_configs()
device = torch.device("cuda") if torch.cuda.is_available() else torch.device("cpu")
inference_pipelines = {}
def preload_all_resources():
print("Preloading resources...")
setup_configs()
global downloaded_content_style_tokenizer_path, downloaded_fmt_path, downloaded_vocoder_path
if not downloaded_resources["tokenizer_vq8192"]:
downloaded_content_style_tokenizer_path = snapshot_download(repo_id="amphion/Vevo", repo_type="model", cache_dir="./ckpts/Vevo", allow_patterns=["tokenizer/vq8192/*"])
downloaded_resources["tokenizer_vq8192"] = True
if not downloaded_resources["fmt_Vq8192ToMels"]:
downloaded_fmt_path = snapshot_download(repo_id="amphion/Vevo", repo_type="model", cache_dir="./ckpts/Vevo", allow_patterns=["acoustic_modeling/Vq8192ToMels/*"])
downloaded_resources["fmt_Vq8192ToMels"] = True
if not downloaded_resources["vocoder"]:
downloaded_vocoder_path = snapshot_download(repo_id="amphion/Vevo", repo_type="model", cache_dir="./ckpts/Vevo", allow_patterns=["acoustic_modeling/Vocoder/*"])
downloaded_resources["vocoder"] = True
print("Resources ready.")
downloaded_content_style_tokenizer_path = None
downloaded_fmt_path = None
downloaded_vocoder_path = None
preload_all_resources()
def get_pipeline():
if "timbre" in inference_pipelines: return inference_pipelines["timbre"]
pipeline = VevoInferencePipeline(
content_style_tokenizer_ckpt_path=os.path.join(downloaded_content_style_tokenizer_path, "tokenizer/vq8192"),
fmt_cfg_path="./models/vc/vevo/config/Vq8192ToMels.json",
fmt_ckpt_path=os.path.join(downloaded_fmt_path, "acoustic_modeling/Vq8192ToMels"),
vocoder_cfg_path="./models/vc/vevo/config/Vocoder.json",
vocoder_ckpt_path=os.path.join(downloaded_vocoder_path, "acoustic_modeling/Vocoder"),
device=device,
)
inference_pipelines["timbre"] = pipeline
return pipeline
@spaces.GPU()
def vevo_timbre(content_wav, reference_wav):
session_id = str(uuid.uuid4())[:8]
temp_content_path = f"wav/c_{session_id}.wav"
temp_reference_path = f"wav/r_{session_id}.wav"
output_path = f"wav/out_{session_id}.wav"
if content_wav is None or reference_wav is None:
raise ValueError("Please upload audio files")
try:
# --- 1. پردازش ورودی ---
if isinstance(content_wav, tuple):
content_sr, content_data = content_wav if isinstance(content_wav[0], int) else (content_wav[1], content_wav[0])
else:
content_sr, content_data = content_wav
if len(content_data.shape) > 1: content_data = np.mean(content_data, axis=1)
content_tensor = torch.FloatTensor(content_data).unsqueeze(0)
if content_sr != 24000:
content_tensor = torchaudio.functional.resample(content_tensor, content_sr, 24000)
content_tensor = content_tensor / (torch.max(torch.abs(content_tensor)) + 1e-6) * 0.95
content_full_np = content_tensor.squeeze().numpy()
# --- 2. پردازش رفرنس ---
if isinstance(reference_wav, tuple):
ref_sr, ref_data = reference_wav if isinstance(reference_wav[0], int) else (reference_wav[1], reference_wav[0])
else:
ref_sr, ref_data = reference_wav
if len(ref_data.shape) > 1: ref_data = np.mean(ref_data, axis=1)
ref_tensor = torch.FloatTensor(ref_data).unsqueeze(0)
if ref_sr != 24000:
ref_tensor = torchaudio.functional.resample(ref_tensor, ref_sr, 24000)
ref_tensor = ref_tensor / (torch.max(torch.abs(ref_tensor)) + 1e-6) * 0.95
if ref_tensor.shape[1] > 24000 * 20: ref_tensor = ref_tensor[:, :24000 * 20]
save_audio_pcm16(ref_tensor, temp_reference_path, 24000)
# --- 3. منطق پردازش با استفاده از PyDub ---
pipeline = get_pipeline()
SR = 24000
NEW_CHUNK_SEC = 10.0
CONTEXT_SEC = 3.0
new_chunk_samples = int(NEW_CHUNK_SEC * SR)
context_samples = int(CONTEXT_SEC * SR)
total_samples = len(content_full_np)
# ایجاد یک AudioSegment خالی برای جمعآوری خروجی نهایی
final_audio_segment = AudioSegment.empty()
current_cursor = 0
print(f"[{session_id}] Processing with PyDub stitching...")
while current_cursor < total_samples:
start_slice = max(0, current_cursor - context_samples)
end_slice = min(total_samples, current_cursor + new_chunk_samples)
if start_slice >= end_slice: break
chunk_np = content_full_np[start_slice:end_slice]
save_audio_pcm16(torch.FloatTensor(chunk_np).unsqueeze(0), temp_content_path, SR)
try:
gen = pipeline.inference_fm(
src_wav_path=temp_content_path,
timbre_ref_wav_path=temp_reference_path,
flow_matching_steps=64,
)
if torch.isnan(gen).any(): gen = torch.nan_to_num(gen, nan=0.0)
gen_np = gen.detach().cpu().squeeze().numpy()
# محاسبه مقدار برشی (حذف کانتکست تکراری)
trim_samples = current_cursor - start_slice
if len(gen_np) > trim_samples:
valid_part_np = gen_np[trim_samples:]
# تبدیل به فرمت PyDub
new_segment = numpy_to_audiosegment(valid_part_np, SR)
# اتصال:
# اگر اولین تکه نیست، یک فید (Crossfade) بسیار کوتاه (5 میلی ثانیه)
# اعمال میکنیم تا صدای "تیک" حذف شود.
if len(final_audio_segment) > 0:
# تکنیک: یک فید بسیار ریز (Crossfade 5ms)
# نکته: PyDub برای کراسفید نیاز به همپوشانی دارد، اما چون ما کانتکست را دقیق بریدیم،
# اینجا از append ساده استفاده میکنیم و فقط لبهها را نرم میکنیم.
# نرم کردن ابتدای تکه جدید (Fade In 5ms)
new_segment = new_segment.fade_in(5)
# نرم کردن انتهای تکه قبلی (Fade Out 5ms) - (قبلاً انجام شده یا الان انجام میدیم)
# در اینجا فقط چسباندن (Append) با فید این کافیست.
final_audio_segment += new_segment
else:
final_audio_segment += new_segment
current_cursor = end_slice
except Exception as e:
print(f"Error: {e}")
current_cursor = end_slice # Skip on error
# ذخیره خروجی نهایی با PyDub
final_audio_segment.export(output_path, format="wav")
return output_path
finally:
if os.path.exists(temp_content_path): os.remove(temp_content_path)
if os.path.exists(temp_reference_path): os.remove(temp_reference_path)
with gr.Blocks(title="Vevo-Timbre (PyDub)") as demo:
gr.Markdown("## Vevo-Timbre: Voice Conversion")
gr.Markdown("Seamless stitching powered by PyDub library.")
with gr.Row():
with gr.Column():
timbre_content = gr.Audio(label="Source Audio", type="numpy")
timbre_reference = gr.Audio(label="Target Timbre", type="numpy")
timbre_button = gr.Button("Generate", variant="primary")
with gr.Column():
timbre_output = gr.Audio(label="Result")
timbre_button.click(vevo_timbre, inputs=[timbre_content, timbre_reference], outputs=timbre_output)
demo.launch() |