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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 | |
| # 创建一个全局变量来跟踪已下载的资源 | |
| # Create a global variable to track downloaded resources | |
| downloaded_resources = { | |
| "configs": False, | |
| "tokenizer_vq32": False, | |
| "tokenizer_vq8192": False, | |
| "ar_Vq32ToVq8192": False, | |
| "ar_PhoneToVq8192": False, | |
| "fmt_Vq8192ToMels": False, | |
| "vocoder": False | |
| } | |
| def install_espeak(): | |
| """Detect and install espeak-ng dependency""" | |
| try: | |
| # Check if espeak-ng is already installed | |
| result = subprocess.run(["which", "espeak-ng"], capture_output=True, text=True) | |
| if result.returncode != 0: | |
| print("Detected espeak-ng not installed in the system, attempting to install...") | |
| # Try to install espeak-ng and its data using apt-get | |
| subprocess.run(["apt-get", "update"], check=True) | |
| # Install espeak-ng and the corresponding language data package | |
| subprocess.run(["apt-get", "install", "-y", "espeak-ng", "espeak-ng-data"], check=True) | |
| print("espeak-ng and its data packages installed successfully!") | |
| else: | |
| print("espeak-ng is already installed in the system.") | |
| # Even if already installed, try to update data to ensure integrity (optional but sometimes helpful) | |
| # print("Attempting to update espeak-ng data...") | |
| # subprocess.run(["apt-get", "update"], check=True) | |
| # subprocess.run(["apt-get", "install", "--only-upgrade", "-y", "espeak-ng-data"], check=True) | |
| # Verify Chinese support (optional) | |
| try: | |
| voices_result = subprocess.run(["espeak-ng", "--voices=cmn"], capture_output=True, text=True, check=True) | |
| if "cmn" in voices_result.stdout: | |
| print("espeak-ng supports 'cmn' language.") | |
| else: | |
| print("Warning: espeak-ng is installed, but 'cmn' language still seems unavailable.") | |
| except Exception as e: | |
| print(f"Error verifying espeak-ng Chinese support (may not affect functionality): {e}") | |
| except Exception as e: | |
| print(f"Error installing espeak-ng: {e}") | |
| print("Please try to run manually: apt-get update && apt-get install -y espeak-ng espeak-ng-data") | |
| # Install espeak before all other operations | |
| install_espeak() | |
| def patch_langsegment_init(): | |
| try: | |
| # Try to find the location of the LangSegment package | |
| spec = importlib.util.find_spec("LangSegment") | |
| if spec is None or spec.origin is None: | |
| print("Unable to locate LangSegment package.") | |
| return | |
| # Build the path to __init__.py | |
| init_path = os.path.join(os.path.dirname(spec.origin), '__init__.py') | |
| if not os.path.exists(init_path): | |
| print(f"LangSegment __init__.py file not found at: {init_path}") | |
| # Try to find in site-packages, applicable in some environments | |
| for site_pkg_path in site.getsitepackages(): | |
| potential_path = os.path.join(site_pkg_path, 'LangSegment', '__init__.py') | |
| if os.path.exists(potential_path): | |
| init_path = potential_path | |
| print(f"Found __init__.py in site-packages: {init_path}") | |
| break | |
| else: # If the loop ends normally (no break) | |
| print(f"Also unable to find __init__.py in site-packages") | |
| return | |
| print(f"Attempting to read LangSegment __init__.py: {init_path}") | |
| with open(init_path, 'r') as f: | |
| lines = f.readlines() | |
| modified = False | |
| new_lines = [] | |
| target_line_prefix = "from .LangSegment import" | |
| for line in lines: | |
| stripped_line = line.strip() | |
| if stripped_line.startswith(target_line_prefix): | |
| if 'setLangfilters' in stripped_line or 'getLangfilters' in stripped_line: | |
| print(f"Found line that needs modification: {stripped_line}") | |
| # Remove setLangfilters and getLangfilters | |
| modified_line = stripped_line.replace(',setLangfilters', '') | |
| modified_line = modified_line.replace(',getLangfilters', '') | |
| # Ensure comma handling is correct (e.g., if they are the last items) | |
| modified_line = modified_line.replace('setLangfilters,', '') | |
| modified_line = modified_line.replace('getLangfilters,', '') | |
| # If they are the only extra imports, remove any redundant commas | |
| modified_line = modified_line.rstrip(',') | |
| new_lines.append(modified_line + '\n') | |
| modified = True | |
| print(f"Modified line: {modified_line.strip()}") | |
| else: | |
| new_lines.append(line) # Line is fine, keep as is | |
| else: | |
| new_lines.append(line) # Non-target line, keep as is | |
| if modified: | |
| print(f"Attempting to write back modified LangSegment __init__.py to: {init_path}") | |
| try: | |
| with open(init_path, 'w') as f: | |
| f.writelines(new_lines) | |
| print("LangSegment __init__.py modified successfully.") | |
| # Try to reload the module to make changes effective (may not work, depending on import chain) | |
| try: | |
| import LangSegment | |
| importlib.reload(LangSegment) | |
| print("LangSegment module has been attempted to reload.") | |
| except Exception as reload_e: | |
| print(f"Error reloading LangSegment (may have no impact): {reload_e}") | |
| except PermissionError: | |
| print(f"Error: Insufficient permissions to modify {init_path}. Consider modifying requirements.txt.") | |
| except Exception as write_e: | |
| print(f"Other error occurred when writing LangSegment __init__.py: {write_e}") | |
| else: | |
| print("LangSegment __init__.py doesn't need modification.") | |
| except ImportError: | |
| print("LangSegment package not found, unable to fix.") | |
| except Exception as e: | |
| print(f"Unexpected error occurred when fixing LangSegment package: {e}") | |
| # Execute the fix before all other imports (especially Amphion) that might trigger LangSegment | |
| patch_langsegment_init() | |
| # Clone Amphion repository | |
| if not os.path.exists("Amphion"): | |
| subprocess.run(["git", "clone", "https://github.com/open-mmlab/Amphion.git"]) | |
| os.chdir("Amphion") | |
| else: | |
| if not os.getcwd().endswith("Amphion"): | |
| os.chdir("Amphion") | |
| # Add Amphion to the path | |
| if os.path.dirname(os.path.abspath("Amphion")) not in sys.path: | |
| sys.path.append(os.path.dirname(os.path.abspath("Amphion"))) | |
| # Ensure needed directories exist | |
| os.makedirs("wav", exist_ok=True) | |
| os.makedirs("ckpts/Vevo", exist_ok=True) | |
| from models.vc.vevo.vevo_utils import VevoInferencePipeline, save_audio, load_wav | |
| # Download and setup config files | |
| def setup_configs(): | |
| if downloaded_resources["configs"]: | |
| print("Config files already downloaded, skipping...") | |
| return | |
| config_path = "models/vc/vevo/config" | |
| os.makedirs(config_path, exist_ok=True) | |
| config_files = [ | |
| "PhoneToVq8192.json", | |
| "Vocoder.json", | |
| "Vq32ToVq8192.json", | |
| "Vq8192ToMels.json", | |
| "hubert_large_l18_c32.yaml", | |
| ] | |
| 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", | |
| ) | |
| os.makedirs(os.path.dirname(file_path), exist_ok=True) | |
| # Copy file to target location | |
| subprocess.run(["cp", file_data, file_path]) | |
| except Exception as e: | |
| print(f"Error downloading config file {file}: {e}") | |
| downloaded_resources["configs"] = True | |
| setup_configs() | |
| # Device configuration | |
| device = torch.device("cuda") if torch.cuda.is_available() else torch.device("cpu") | |
| print(f"Using device: {device}") | |
| # Initialize pipeline dictionary | |
| inference_pipelines = {} | |
| # Download all necessary model resources at startup | |
| def preload_all_resources(): | |
| print("Preloading all model resources...") | |
| # Download configuration files | |
| setup_configs() | |
| # Store the downloaded model paths | |
| global downloaded_content_tokenizer_path | |
| global downloaded_content_style_tokenizer_path | |
| global downloaded_ar_vq32_path | |
| global downloaded_ar_phone_path | |
| global downloaded_fmt_path | |
| global downloaded_vocoder_path | |
| # Download Content Tokenizer (vq32) | |
| if not downloaded_resources["tokenizer_vq32"]: | |
| print("Preloading Content Tokenizer (vq32)...") | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["tokenizer/vq32/*"], | |
| ) | |
| downloaded_content_tokenizer_path = local_dir | |
| downloaded_resources["tokenizer_vq32"] = True | |
| print("Content Tokenizer (vq32) download completed") | |
| # Download Content-Style Tokenizer (vq8192) | |
| if not downloaded_resources["tokenizer_vq8192"]: | |
| print("Preloading Content-Style Tokenizer (vq8192)...") | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["tokenizer/vq8192/*"], | |
| ) | |
| downloaded_content_style_tokenizer_path = local_dir | |
| downloaded_resources["tokenizer_vq8192"] = True | |
| print("Content-Style Tokenizer (vq8192) download completed") | |
| # Download Autoregressive Transformer (Vq32ToVq8192) | |
| if not downloaded_resources["ar_Vq32ToVq8192"]: | |
| print("Preloading Autoregressive Transformer (Vq32ToVq8192)...") | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["contentstyle_modeling/Vq32ToVq8192/*"], | |
| ) | |
| downloaded_ar_vq32_path = local_dir | |
| downloaded_resources["ar_Vq32ToVq8192"] = True | |
| print("Autoregressive Transformer (Vq32ToVq8192) download completed") | |
| # Download Autoregressive Transformer (PhoneToVq8192) | |
| if not downloaded_resources["ar_PhoneToVq8192"]: | |
| print("Preloading Autoregressive Transformer (PhoneToVq8192)...") | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["contentstyle_modeling/PhoneToVq8192/*"], | |
| ) | |
| downloaded_ar_phone_path = local_dir | |
| downloaded_resources["ar_PhoneToVq8192"] = True | |
| print("Autoregressive Transformer (PhoneToVq8192) download completed") | |
| # Download Flow Matching Transformer | |
| if not downloaded_resources["fmt_Vq8192ToMels"]: | |
| print("Preloading Flow Matching Transformer (Vq8192ToMels)...") | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vq8192ToMels/*"], | |
| ) | |
| downloaded_fmt_path = local_dir | |
| downloaded_resources["fmt_Vq8192ToMels"] = True | |
| print("Flow Matching Transformer (Vq8192ToMels) download completed") | |
| # Download Vocoder | |
| if not downloaded_resources["vocoder"]: | |
| print("Preloading Vocoder...") | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vocoder/*"], | |
| ) | |
| downloaded_vocoder_path = local_dir | |
| downloaded_resources["vocoder"] = True | |
| print("Vocoder download completed") | |
| print("All model resources preloading completed!") | |
| # Initialize path variables to store downloaded model paths | |
| downloaded_content_tokenizer_path = None | |
| downloaded_content_style_tokenizer_path = None | |
| downloaded_ar_vq32_path = None | |
| downloaded_ar_phone_path = None | |
| downloaded_fmt_path = None | |
| downloaded_vocoder_path = None | |
| # Preload all resources before creating the Gradio interface | |
| preload_all_resources() | |
| def get_pipeline(pipeline_type): | |
| if pipeline_type in inference_pipelines: | |
| return inference_pipelines[pipeline_type] | |
| # Initialize pipeline based on the required pipeline type | |
| if pipeline_type == "style" or pipeline_type == "voice": | |
| # Use already downloaded Content Tokenizer | |
| if downloaded_resources["tokenizer_vq32"]: | |
| content_tokenizer_ckpt_path = os.path.join( | |
| downloaded_content_tokenizer_path, "tokenizer/vq32/hubert_large_l18_c32.pkl" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["tokenizer/vq32/*"], | |
| ) | |
| content_tokenizer_ckpt_path = os.path.join( | |
| local_dir, "tokenizer/vq32/hubert_large_l18_c32.pkl" | |
| ) | |
| # Use already downloaded Content-Style Tokenizer | |
| if downloaded_resources["tokenizer_vq8192"]: | |
| content_style_tokenizer_ckpt_path = os.path.join( | |
| downloaded_content_style_tokenizer_path, "tokenizer/vq8192" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["tokenizer/vq8192/*"], | |
| ) | |
| content_style_tokenizer_ckpt_path = os.path.join(local_dir, "tokenizer/vq8192") | |
| # Use already downloaded Autoregressive Transformer | |
| ar_cfg_path = "./models/vc/vevo/config/Vq32ToVq8192.json" | |
| if downloaded_resources["ar_Vq32ToVq8192"]: | |
| ar_ckpt_path = os.path.join( | |
| downloaded_ar_vq32_path, "contentstyle_modeling/Vq32ToVq8192" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["contentstyle_modeling/Vq32ToVq8192/*"], | |
| ) | |
| ar_ckpt_path = os.path.join(local_dir, "contentstyle_modeling/Vq32ToVq8192") | |
| # Use already downloaded Flow Matching Transformer | |
| fmt_cfg_path = "./models/vc/vevo/config/Vq8192ToMels.json" | |
| if downloaded_resources["fmt_Vq8192ToMels"]: | |
| fmt_ckpt_path = os.path.join( | |
| downloaded_fmt_path, "acoustic_modeling/Vq8192ToMels" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vq8192ToMels/*"], | |
| ) | |
| fmt_ckpt_path = os.path.join(local_dir, "acoustic_modeling/Vq8192ToMels") | |
| # Use already downloaded Vocoder | |
| vocoder_cfg_path = "./models/vc/vevo/config/Vocoder.json" | |
| if downloaded_resources["vocoder"]: | |
| vocoder_ckpt_path = os.path.join( | |
| downloaded_vocoder_path, "acoustic_modeling/Vocoder" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vocoder/*"], | |
| ) | |
| vocoder_ckpt_path = os.path.join(local_dir, "acoustic_modeling/Vocoder") | |
| # Initialize pipeline | |
| inference_pipeline = VevoInferencePipeline( | |
| content_tokenizer_ckpt_path=content_tokenizer_ckpt_path, | |
| content_style_tokenizer_ckpt_path=content_style_tokenizer_ckpt_path, | |
| ar_cfg_path=ar_cfg_path, | |
| ar_ckpt_path=ar_ckpt_path, | |
| fmt_cfg_path=fmt_cfg_path, | |
| fmt_ckpt_path=fmt_ckpt_path, | |
| vocoder_cfg_path=vocoder_cfg_path, | |
| vocoder_ckpt_path=vocoder_ckpt_path, | |
| device=device, | |
| ) | |
| elif pipeline_type == "timbre": | |
| # Use already downloaded Content-Style Tokenizer | |
| if downloaded_resources["tokenizer_vq8192"]: | |
| content_style_tokenizer_ckpt_path = os.path.join( | |
| downloaded_content_style_tokenizer_path, "tokenizer/vq8192" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["tokenizer/vq8192/*"], | |
| ) | |
| content_style_tokenizer_ckpt_path = os.path.join(local_dir, "tokenizer/vq8192") | |
| # Use already downloaded Flow Matching Transformer | |
| fmt_cfg_path = "./models/vc/vevo/config/Vq8192ToMels.json" | |
| if downloaded_resources["fmt_Vq8192ToMels"]: | |
| fmt_ckpt_path = os.path.join( | |
| downloaded_fmt_path, "acoustic_modeling/Vq8192ToMels" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vq8192ToMels/*"], | |
| ) | |
| fmt_ckpt_path = os.path.join(local_dir, "acoustic_modeling/Vq8192ToMels") | |
| # Use already downloaded Vocoder | |
| vocoder_cfg_path = "./models/vc/vevo/config/Vocoder.json" | |
| if downloaded_resources["vocoder"]: | |
| vocoder_ckpt_path = os.path.join( | |
| downloaded_vocoder_path, "acoustic_modeling/Vocoder" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vocoder/*"], | |
| ) | |
| vocoder_ckpt_path = os.path.join(local_dir, "acoustic_modeling/Vocoder") | |
| # Initialize pipeline | |
| inference_pipeline = VevoInferencePipeline( | |
| content_style_tokenizer_ckpt_path=content_style_tokenizer_ckpt_path, | |
| fmt_cfg_path=fmt_cfg_path, | |
| fmt_ckpt_path=fmt_ckpt_path, | |
| vocoder_cfg_path=vocoder_cfg_path, | |
| vocoder_ckpt_path=vocoder_ckpt_path, | |
| device=device, | |
| ) | |
| elif pipeline_type == "tts": | |
| # Use already downloaded Content-Style Tokenizer | |
| if downloaded_resources["tokenizer_vq8192"]: | |
| content_style_tokenizer_ckpt_path = os.path.join( | |
| downloaded_content_style_tokenizer_path, "tokenizer/vq8192" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["tokenizer/vq8192/*"], | |
| ) | |
| content_style_tokenizer_ckpt_path = os.path.join(local_dir, "tokenizer/vq8192") | |
| # Use already downloaded Autoregressive Transformer (TTS specific) | |
| ar_cfg_path = "./models/vc/vevo/config/PhoneToVq8192.json" | |
| if downloaded_resources["ar_PhoneToVq8192"]: | |
| ar_ckpt_path = os.path.join( | |
| downloaded_ar_phone_path, "contentstyle_modeling/PhoneToVq8192" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["contentstyle_modeling/PhoneToVq8192/*"], | |
| ) | |
| ar_ckpt_path = os.path.join(local_dir, "contentstyle_modeling/PhoneToVq8192") | |
| # Use already downloaded Flow Matching Transformer | |
| fmt_cfg_path = "./models/vc/vevo/config/Vq8192ToMels.json" | |
| if downloaded_resources["fmt_Vq8192ToMels"]: | |
| fmt_ckpt_path = os.path.join( | |
| downloaded_fmt_path, "acoustic_modeling/Vq8192ToMels" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vq8192ToMels/*"], | |
| ) | |
| fmt_ckpt_path = os.path.join(local_dir, "acoustic_modeling/Vq8192ToMels") | |
| # Use already downloaded Vocoder | |
| vocoder_cfg_path = "./models/vc/vevo/config/Vocoder.json" | |
| if downloaded_resources["vocoder"]: | |
| vocoder_ckpt_path = os.path.join( | |
| downloaded_vocoder_path, "acoustic_modeling/Vocoder" | |
| ) | |
| else: | |
| # Fallback to direct download | |
| local_dir = snapshot_download( | |
| repo_id="amphion/Vevo", | |
| repo_type="model", | |
| cache_dir="./ckpts/Vevo", | |
| allow_patterns=["acoustic_modeling/Vocoder/*"], | |
| ) | |
| vocoder_ckpt_path = os.path.join(local_dir, "acoustic_modeling/Vocoder") | |
| # Initialize pipeline | |
| inference_pipeline = VevoInferencePipeline( | |
| content_style_tokenizer_ckpt_path=content_style_tokenizer_ckpt_path, | |
| ar_cfg_path=ar_cfg_path, | |
| ar_ckpt_path=ar_ckpt_path, | |
| fmt_cfg_path=fmt_cfg_path, | |
| fmt_ckpt_path=fmt_ckpt_path, | |
| vocoder_cfg_path=vocoder_cfg_path, | |
| vocoder_ckpt_path=vocoder_ckpt_path, | |
| device=device, | |
| ) | |
| # Cache pipeline instance | |
| inference_pipelines[pipeline_type] = inference_pipeline | |
| return inference_pipeline | |
| # Implement VEVO functionality functions | |
| def vevo_style(content_wav, style_wav): | |
| temp_content_path = "wav/temp_content.wav" | |
| temp_style_path = "wav/temp_style.wav" | |
| output_path = "wav/output_vevostyle.wav" | |
| # Check and process audio data | |
| if content_wav is None or style_wav is None: | |
| raise ValueError("Please upload audio files") | |
| # Process audio format | |
| if isinstance(content_wav, tuple) and len(content_wav) == 2: | |
| if isinstance(content_wav[0], np.ndarray): | |
| content_data, content_sr = content_wav | |
| else: | |
| content_sr, content_data = content_wav | |
| # Ensure single channel | |
| if len(content_data.shape) > 1 and content_data.shape[1] > 1: | |
| content_data = np.mean(content_data, axis=1) | |
| # Resample to 24kHz | |
| if content_sr != 24000: | |
| content_tensor = torch.FloatTensor(content_data).unsqueeze(0) | |
| content_tensor = torchaudio.functional.resample(content_tensor, content_sr, 24000) | |
| content_sr = 24000 | |
| else: | |
| content_tensor = torch.FloatTensor(content_data).unsqueeze(0) | |
| # Normalize volume | |
| content_tensor = content_tensor / (torch.max(torch.abs(content_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid content audio format") | |
| if isinstance(style_wav[0], np.ndarray): | |
| style_data, style_sr = style_wav | |
| else: | |
| style_sr, style_data = style_wav | |
| # Ensure single channel | |
| if len(style_data.shape) > 1 and style_data.shape[1] > 1: | |
| style_data = np.mean(style_data, axis=1) | |
| # Resample to 24kHz | |
| if style_sr != 24000: | |
| style_tensor = torch.FloatTensor(style_data).unsqueeze(0) | |
| style_tensor = torchaudio.functional.resample(style_tensor, style_sr, 24000) | |
| style_sr = 24000 | |
| else: | |
| style_tensor = torch.FloatTensor(style_data).unsqueeze(0) | |
| # Normalize volume | |
| style_tensor = style_tensor / (torch.max(torch.abs(style_tensor)) + 1e-6) * 0.95 | |
| # Print debug information | |
| print(f"Content audio shape: {content_tensor.shape}, sample rate: {content_sr}") | |
| print(f"Style audio shape: {style_tensor.shape}, sample rate: {style_sr}") | |
| # Save audio | |
| torchaudio.save(temp_content_path, content_tensor, content_sr) | |
| torchaudio.save(temp_style_path, style_tensor, style_sr) | |
| try: | |
| # Get pipeline | |
| pipeline = get_pipeline("style") | |
| # Inference | |
| gen_audio = pipeline.inference_ar_and_fm( | |
| src_wav_path=temp_content_path, | |
| src_text=None, | |
| style_ref_wav_path=temp_style_path, | |
| timbre_ref_wav_path=temp_content_path, | |
| ) | |
| # Check if generated audio is numerical anomaly | |
| if torch.isnan(gen_audio).any() or torch.isinf(gen_audio).any(): | |
| print("Warning: Generated audio contains NaN or Inf values") | |
| gen_audio = torch.nan_to_num(gen_audio, nan=0.0, posinf=0.95, neginf=-0.95) | |
| print(f"Generated audio shape: {gen_audio.shape}, max: {torch.max(gen_audio)}, min: {torch.min(gen_audio)}") | |
| # Save generated audio | |
| save_audio(gen_audio, output_path=output_path) | |
| return output_path | |
| except Exception as e: | |
| print(f"Error during processing: {e}") | |
| import traceback | |
| traceback.print_exc() | |
| raise e | |
| def vevo_timbre(content_wav, reference_wav): | |
| temp_content_path = "wav/temp_content.wav" | |
| temp_reference_path = "wav/temp_reference.wav" | |
| output_path = "wav/output_vevotimbre.wav" | |
| # Check and process audio data | |
| if content_wav is None or reference_wav is None: | |
| raise ValueError("Please upload audio files") | |
| # Process content audio format | |
| if isinstance(content_wav, tuple) and len(content_wav) == 2: | |
| if isinstance(content_wav[0], np.ndarray): | |
| content_data, content_sr = content_wav | |
| else: | |
| content_sr, content_data = content_wav | |
| # Ensure single channel | |
| if len(content_data.shape) > 1 and content_data.shape[1] > 1: | |
| content_data = np.mean(content_data, axis=1) | |
| # Resample to 24kHz | |
| if content_sr != 24000: | |
| content_tensor = torch.FloatTensor(content_data).unsqueeze(0) | |
| content_tensor = torchaudio.functional.resample(content_tensor, content_sr, 24000) | |
| content_sr = 24000 | |
| else: | |
| content_tensor = torch.FloatTensor(content_data).unsqueeze(0) | |
| # Normalize volume | |
| content_tensor = content_tensor / (torch.max(torch.abs(content_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid content audio format") | |
| # Process reference audio format | |
| if isinstance(reference_wav, tuple) and len(reference_wav) == 2: | |
| if isinstance(reference_wav[0], np.ndarray): | |
| reference_data, reference_sr = reference_wav | |
| else: | |
| reference_sr, reference_data = reference_wav | |
| # Ensure single channel | |
| if len(reference_data.shape) > 1 and reference_data.shape[1] > 1: | |
| reference_data = np.mean(reference_data, axis=1) | |
| # Resample to 24kHz | |
| if reference_sr != 24000: | |
| reference_tensor = torch.FloatTensor(reference_data).unsqueeze(0) | |
| reference_tensor = torchaudio.functional.resample(reference_tensor, reference_sr, 24000) | |
| reference_sr = 24000 | |
| else: | |
| reference_tensor = torch.FloatTensor(reference_data).unsqueeze(0) | |
| # Normalize volume | |
| reference_tensor = reference_tensor / (torch.max(torch.abs(reference_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid reference audio format") | |
| # Print debug information | |
| print(f"Content audio shape: {content_tensor.shape}, sample rate: {content_sr}") | |
| print(f"Reference audio shape: {reference_tensor.shape}, sample rate: {reference_sr}") | |
| # Save uploaded audio | |
| torchaudio.save(temp_content_path, content_tensor, content_sr) | |
| torchaudio.save(temp_reference_path, reference_tensor, reference_sr) | |
| try: | |
| # Get pipeline | |
| pipeline = get_pipeline("timbre") | |
| # Inference | |
| gen_audio = pipeline.inference_fm( | |
| src_wav_path=temp_content_path, | |
| timbre_ref_wav_path=temp_reference_path, | |
| flow_matching_steps=32, | |
| ) | |
| # Check if generated audio is numerical anomaly | |
| if torch.isnan(gen_audio).any() or torch.isinf(gen_audio).any(): | |
| print("Warning: Generated audio contains NaN or Inf values") | |
| gen_audio = torch.nan_to_num(gen_audio, nan=0.0, posinf=0.95, neginf=-0.95) | |
| print(f"Generated audio shape: {gen_audio.shape}, max: {torch.max(gen_audio)}, min: {torch.min(gen_audio)}") | |
| # Save generated audio | |
| save_audio(gen_audio, output_path=output_path) | |
| return output_path | |
| except Exception as e: | |
| print(f"Error during processing: {e}") | |
| import traceback | |
| traceback.print_exc() | |
| raise e | |
| def vevo_voice(content_wav, style_reference_wav, timbre_reference_wav): | |
| temp_content_path = "wav/temp_content.wav" | |
| temp_style_path = "wav/temp_style.wav" | |
| temp_timbre_path = "wav/temp_timbre.wav" | |
| output_path = "wav/output_vevovoice.wav" | |
| # Check and process audio data | |
| if content_wav is None or style_reference_wav is None or timbre_reference_wav is None: | |
| raise ValueError("Please upload all required audio files") | |
| # Process content audio format | |
| if isinstance(content_wav, tuple) and len(content_wav) == 2: | |
| if isinstance(content_wav[0], np.ndarray): | |
| content_data, content_sr = content_wav | |
| else: | |
| content_sr, content_data = content_wav | |
| # Ensure single channel | |
| if len(content_data.shape) > 1 and content_data.shape[1] > 1: | |
| content_data = np.mean(content_data, axis=1) | |
| # Resample to 24kHz | |
| if content_sr != 24000: | |
| content_tensor = torch.FloatTensor(content_data).unsqueeze(0) | |
| content_tensor = torchaudio.functional.resample(content_tensor, content_sr, 24000) | |
| content_sr = 24000 | |
| else: | |
| content_tensor = torch.FloatTensor(content_data).unsqueeze(0) | |
| # Normalize volume | |
| content_tensor = content_tensor / (torch.max(torch.abs(content_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid content audio format") | |
| # Process style reference audio format | |
| if isinstance(style_reference_wav, tuple) and len(style_reference_wav) == 2: | |
| if isinstance(style_reference_wav[0], np.ndarray): | |
| style_data, style_sr = style_reference_wav | |
| else: | |
| style_sr, style_data = style_reference_wav | |
| # Ensure single channel | |
| if len(style_data.shape) > 1 and style_data.shape[1] > 1: | |
| style_data = np.mean(style_data, axis=1) | |
| # Resample to 24kHz | |
| if style_sr != 24000: | |
| style_tensor = torch.FloatTensor(style_data).unsqueeze(0) | |
| style_tensor = torchaudio.functional.resample(style_tensor, style_sr, 24000) | |
| style_sr = 24000 | |
| else: | |
| style_tensor = torch.FloatTensor(style_data).unsqueeze(0) | |
| # Normalize volume | |
| style_tensor = style_tensor / (torch.max(torch.abs(style_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid style reference audio format") | |
| # Process timbre reference audio format | |
| if isinstance(timbre_reference_wav, tuple) and len(timbre_reference_wav) == 2: | |
| if isinstance(timbre_reference_wav[0], np.ndarray): | |
| timbre_data, timbre_sr = timbre_reference_wav | |
| else: | |
| timbre_sr, timbre_data = timbre_reference_wav | |
| # Ensure single channel | |
| if len(timbre_data.shape) > 1 and timbre_data.shape[1] > 1: | |
| timbre_data = np.mean(timbre_data, axis=1) | |
| # Resample to 24kHz | |
| if timbre_sr != 24000: | |
| timbre_tensor = torch.FloatTensor(timbre_data).unsqueeze(0) | |
| timbre_tensor = torchaudio.functional.resample(timbre_tensor, timbre_sr, 24000) | |
| timbre_sr = 24000 | |
| else: | |
| timbre_tensor = torch.FloatTensor(timbre_data).unsqueeze(0) | |
| # Normalize volume | |
| timbre_tensor = timbre_tensor / (torch.max(torch.abs(timbre_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid timbre reference audio format") | |
| # Print debug information | |
| print(f"Content audio shape: {content_tensor.shape}, sample rate: {content_sr}") | |
| print(f"Style reference audio shape: {style_tensor.shape}, sample rate: {style_sr}") | |
| print(f"Timbre reference audio shape: {timbre_tensor.shape}, sample rate: {timbre_sr}") | |
| # Save uploaded audio | |
| torchaudio.save(temp_content_path, content_tensor, content_sr) | |
| torchaudio.save(temp_style_path, style_tensor, style_sr) | |
| torchaudio.save(temp_timbre_path, timbre_tensor, timbre_sr) | |
| try: | |
| # Get pipeline | |
| pipeline = get_pipeline("voice") | |
| # Inference | |
| gen_audio = pipeline.inference_ar_and_fm( | |
| src_wav_path=temp_content_path, | |
| src_text=None, | |
| style_ref_wav_path=temp_style_path, | |
| timbre_ref_wav_path=temp_timbre_path, | |
| ) | |
| # Check if generated audio is numerical anomaly | |
| if torch.isnan(gen_audio).any() or torch.isinf(gen_audio).any(): | |
| print("Warning: Generated audio contains NaN or Inf values") | |
| gen_audio = torch.nan_to_num(gen_audio, nan=0.0, posinf=0.95, neginf=-0.95) | |
| print(f"Generated audio shape: {gen_audio.shape}, max: {torch.max(gen_audio)}, min: {torch.min(gen_audio)}") | |
| # Save generated audio | |
| save_audio(gen_audio, output_path=output_path) | |
| return output_path | |
| except Exception as e: | |
| print(f"Error during processing: {e}") | |
| import traceback | |
| traceback.print_exc() | |
| raise e | |
| def vevo_tts(text, ref_wav, timbre_ref_wav=None, style_ref_text=None, src_language="en", ref_language="en", style_ref_text_language="en"): | |
| temp_ref_path = "wav/temp_ref.wav" | |
| temp_timbre_path = "wav/temp_timbre.wav" | |
| output_path = "wav/output_vevotts.wav" | |
| # Check and process audio data | |
| if ref_wav is None: | |
| raise ValueError("Please upload a reference audio file") | |
| # Process reference audio format | |
| if isinstance(ref_wav, tuple) and len(ref_wav) == 2: | |
| if isinstance(ref_wav[0], np.ndarray): | |
| ref_data, ref_sr = ref_wav | |
| else: | |
| ref_sr, ref_data = ref_wav | |
| # Ensure single channel | |
| if len(ref_data.shape) > 1 and ref_data.shape[1] > 1: | |
| ref_data = np.mean(ref_data, axis=1) | |
| # Resample to 24kHz | |
| if ref_sr != 24000: | |
| ref_tensor = torch.FloatTensor(ref_data).unsqueeze(0) | |
| ref_tensor = torchaudio.functional.resample(ref_tensor, ref_sr, 24000) | |
| ref_sr = 24000 | |
| else: | |
| ref_tensor = torch.FloatTensor(ref_data).unsqueeze(0) | |
| # Normalize volume | |
| ref_tensor = ref_tensor / (torch.max(torch.abs(ref_tensor)) + 1e-6) * 0.95 | |
| else: | |
| raise ValueError("Invalid reference audio format") | |
| # Print debug information | |
| print(f"Reference audio shape: {ref_tensor.shape}, sample rate: {ref_sr}") | |
| if style_ref_text: | |
| print(f"Style reference text: {style_ref_text}, language: {style_ref_text_language}") | |
| # Save uploaded audio | |
| torchaudio.save(temp_ref_path, ref_tensor, ref_sr) | |
| if timbre_ref_wav is not None: | |
| if isinstance(timbre_ref_wav, tuple) and len(timbre_ref_wav) == 2: | |
| if isinstance(timbre_ref_wav[0], np.ndarray): | |
| timbre_data, timbre_sr = timbre_ref_wav | |
| else: | |
| timbre_sr, timbre_data = timbre_ref_wav | |
| # Ensure single channel | |
| if len(timbre_data.shape) > 1 and timbre_data.shape[1] > 1: | |
| timbre_data = np.mean(timbre_data, axis=1) | |
| # Resample to 24kHz | |
| if timbre_sr != 24000: | |
| timbre_tensor = torch.FloatTensor(timbre_data).unsqueeze(0) | |
| timbre_tensor = torchaudio.functional.resample(timbre_tensor, timbre_sr, 24000) | |
| timbre_sr = 24000 | |
| else: | |
| timbre_tensor = torch.FloatTensor(timbre_data).unsqueeze(0) | |
| # Normalize volume | |
| timbre_tensor = timbre_tensor / (torch.max(torch.abs(timbre_tensor)) + 1e-6) * 0.95 | |
| print(f"Timbre reference audio shape: {timbre_tensor.shape}, sample rate: {timbre_sr}") | |
| torchaudio.save(temp_timbre_path, timbre_tensor, timbre_sr) | |
| else: | |
| raise ValueError("Invalid timbre reference audio format") | |
| else: | |
| temp_timbre_path = temp_ref_path | |
| try: | |
| # Get pipeline | |
| pipeline = get_pipeline("tts") | |
| # Inference | |
| gen_audio = pipeline.inference_ar_and_fm( | |
| src_wav_path=None, | |
| src_text=text, | |
| style_ref_wav_path=temp_ref_path, | |
| timbre_ref_wav_path=temp_timbre_path, | |
| style_ref_wav_text=style_ref_text, | |
| src_text_language=src_language, | |
| style_ref_wav_text_language=style_ref_text_language, | |
| ) | |
| # Check if generated audio is numerical anomaly | |
| if torch.isnan(gen_audio).any() or torch.isinf(gen_audio).any(): | |
| print("Warning: Generated audio contains NaN or Inf values") | |
| gen_audio = torch.nan_to_num(gen_audio, nan=0.0, posinf=0.95, neginf=-0.95) | |
| print(f"Generated audio shape: {gen_audio.shape}, max: {torch.max(gen_audio)}, min: {torch.min(gen_audio)}") | |
| # Save generated audio | |
| save_audio(gen_audio, output_path=output_path) | |
| return output_path | |
| except Exception as e: | |
| print(f"Error during processing: {e}") | |
| import traceback | |
| traceback.print_exc() | |
| raise e | |
| # Create Gradio interface | |
| with gr.Blocks(title="Vevo: Controllable Zero-Shot Voice Imitation with Self-Supervised Disentanglement") as demo: | |
| gr.Markdown("# Vevo: Controllable Zero-Shot Voice Imitation with Self-Supervised Disentanglement") | |
| # Add link tag line | |
| with gr.Row(elem_id="links_row"): | |
| gr.HTML(""" | |
| <div style="display: flex; justify-content: flex-start; gap: 8px; margin: 0 0; padding-left: 0px;"> | |
| <a href="https://arxiv.org/abs/2502.07243" target="_blank" style="text-decoration: none;"> | |
| <img alt="arXiv Paper" src="https://img.shields.io/badge/arXiv-Paper-red"> | |
| </a> | |
| <a href="https://openreview.net/pdf?id=anQDiQZhDP" target="_blank" style="text-decoration: none;"> | |
| <img alt="ICLR Paper" src="https://img.shields.io/badge/ICLR-Paper-64b63a"> | |
| </a> | |
| <a href="https://huggingface.co./amphion/Vevo" target="_blank" style="text-decoration: none;"> | |
| <img alt="HuggingFace Model" src="https://img.shields.io/badge/%F0%9F%A4%97%20HuggingFace-Model-yellow"> | |
| </a> | |
| <a href="https://github.com/open-mmlab/Amphion/tree/main/models/vc/vevo" target="_blank" style="text-decoration: none;"> | |
| <img alt="GitHub Repo" src="https://img.shields.io/badge/GitHub-Repo-blue"> | |
| </a> | |
| </div> | |
| """) | |
| with gr.Tab("Vevo-Timbre"): | |
| gr.Markdown("### Vevo-Timbre: Maintain style but transfer timbre") | |
| with gr.Row(): | |
| with gr.Column(): | |
| timbre_content = gr.Audio(label="Source Audio", type="numpy") | |
| timbre_reference = gr.Audio(label="Timbre Reference", type="numpy") | |
| timbre_button = gr.Button("Generate") | |
| with gr.Column(): | |
| timbre_output = gr.Audio(label="Result") | |
| timbre_button.click(vevo_timbre, inputs=[timbre_content, timbre_reference], outputs=timbre_output) | |
| with gr.Tab("Vevo-Style"): | |
| gr.Markdown("### Vevo-Style: Maintain timbre but transfer style (accent, emotion, etc.)") | |
| with gr.Row(): | |
| with gr.Column(): | |
| style_content = gr.Audio(label="Source Audio", type="numpy") | |
| style_reference = gr.Audio(label="Style Reference", type="numpy") | |
| style_button = gr.Button("Generate") | |
| with gr.Column(): | |
| style_output = gr.Audio(label="Result") | |
| style_button.click(vevo_style, inputs=[style_content, style_reference], outputs=style_output) | |
| with gr.Tab("Vevo-Voice"): | |
| gr.Markdown("### Vevo-Voice: Transfers both style and timbre with separate references") | |
| with gr.Row(): | |
| with gr.Column(): | |
| voice_content = gr.Audio(label="Source Audio", type="numpy") | |
| voice_style_reference = gr.Audio(label="Style Reference", type="numpy") | |
| voice_timbre_reference = gr.Audio(label="Timbre Reference", type="numpy") | |
| voice_button = gr.Button("Generate") | |
| with gr.Column(): | |
| voice_output = gr.Audio(label="Result") | |
| voice_button.click(vevo_voice, inputs=[voice_content, voice_style_reference, voice_timbre_reference], outputs=voice_output) | |
| with gr.Tab("Vevo-TTS"): | |
| gr.Markdown("### Vevo-TTS: Text-to-speech with separate style and timbre references") | |
| with gr.Row(): | |
| with gr.Column(): | |
| tts_text = gr.Textbox(label="Target Text", placeholder="Enter text to synthesize...", lines=3) | |
| tts_src_language = gr.Dropdown(["en", "zh", "de", "fr", "ja", "ko"], label="Text Language", value="en") | |
| tts_reference = gr.Audio(label="Style Reference", type="numpy") | |
| tts_style_ref_text = gr.Textbox(label="Style Reference Text", placeholder="Enter style reference text...", lines=3) | |
| tts_style_ref_text_language = gr.Dropdown(["en", "zh", "de", "fr", "ja", "ko"], label="Style Reference Text Language", value="en") | |
| tts_timbre_reference = gr.Audio(label="Timbre Reference", type="numpy") | |
| tts_button = gr.Button("Generate") | |
| with gr.Column(): | |
| tts_output = gr.Audio(label="Result") | |
| tts_button.click( | |
| vevo_tts, | |
| inputs=[tts_text, tts_reference, tts_timbre_reference, tts_style_ref_text, tts_src_language, tts_style_ref_text_language], | |
| outputs=tts_output | |
| ) | |
| gr.Markdown(""" | |
| ## About VEVO | |
| VEVO is a versatile voice synthesis and conversion model that offers four main functionalities: | |
| 1. **Vevo-Style**: Maintains timbre but transfers style (accent, emotion, etc.) | |
| 2. **Vevo-Timbre**: Maintains style but transfers timbre | |
| 3. **Vevo-Voice**: Transfers both style and timbre with separate references | |
| 4. **Vevo-TTS**: Text-to-speech with separate style and timbre references | |
| For more information, visit the [Amphion project](https://github.com/open-mmlab/Amphion) | |
| """) | |
| # Launch application | |
| demo.launch() |