app.py

import gradio as gr
import numpy as np
import soundfile as sf
import spaces
import torch
import torchaudio
import librosa
import yaml
import tempfile
import os

from huggingface_hub import hf_hub_download
from transformers import AutoFeatureExtractor, WhisperModel
from torch.nn.utils import parametrizations
from modules.commons import build_model, load_checkpoint, recursive_munch
from modules.campplus.DTDNN import CAMPPlus
from modules.bigvgan import bigvgan
from modules.rmvpe import RMVPE
from modules.audio import mel_spectrogram

# ----------------------------
# Optimization Settings
# ----------------------------

# Set the number of threads to the number of CPU cores
torch.set_num_threads(os.cpu_count())

# Enable optimized backends
torch.backends.openmp.enabled = True
torch.backends.mkldnn.enabled = True
torch.backends.cudnn.enabled = False
torch.backends.cuda.enabled = False

torch.set_grad_enabled(False)

# Force CPU usage
device = torch.device("cpu")
print(f"[DEVICE] | Using device: {device}")

# ----------------------------
# Load Models and Configuration
# ----------------------------

def load_custom_model_from_hf(repo_id, model_filename="pytorch_model.bin", config_filename="config.yml"):
    os.makedirs("./checkpoints", exist_ok=True)
    model_path = hf_hub_download(repo_id=repo_id, filename=model_filename, cache_dir="./checkpoints")
    if config_filename is None:
        return model_path
    config_path = hf_hub_download(repo_id=repo_id, filename=config_filename, cache_dir="./checkpoints")

    return model_path, config_path
    
# Load DiT model
dit_checkpoint_path, dit_config_path = load_custom_model_from_hf(
    "Plachta/Seed-VC",
    "DiT_seed_v2_uvit_whisper_small_wavenet_bigvgan_pruned.pth",
    "config_dit_mel_seed_uvit_whisper_small_wavenet.yml"
)
config = yaml.safe_load(open(dit_config_path, 'r'))
model_params = recursive_munch(config['model_params'])
model = build_model(model_params, stage='DiT')

# Debug: Print model keys to identify correct key
print(f"[INFO] | Model keys: {model.keys()}")

hop_length = config['preprocess_params']['spect_params']['hop_length']
sr = config['preprocess_params']['sr']

# Load DiT checkpoints
model, _, _, _ = load_checkpoint(model, None, dit_checkpoint_path, load_only_params=True, ignore_modules=[], is_distributed=False)
for key in model:
    model[key].eval()
    model[key].to(device)
print("[INFO] | DiT model loaded and set to eval mode.")

model.cfm.estimator.setup_caches(max_batch_size=1, max_seq_length=8192)

# Ensure 'CAMPPlus' is correctly imported and defined
try:
    campplus_model = CAMPPlus(feat_dim=80, embedding_size=192)
    print("[INFO] | CAMPPlus model instantiated.")
except NameError:
    print("[ERROR] | CAMPPlus is not defined. Please check the import path and ensure CAMPPlus is correctly defined.")
    raise

# Set weights_only=True for security
campplus_ckpt_path = load_custom_model_from_hf("funasr/campplus", "campplus_cn_common.bin", config_filename=None)
campplus_state = torch.load(campplus_ckpt_path, map_location="cpu", weights_only=True)
campplus_model.load_state_dict(campplus_state)
campplus_model.eval()
campplus_model.to(device)
print("[INFO] | CAMPPlus model loaded, set to eval mode, and moved to CPU.")

# Load BigVGAN model
bigvgan_model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_22khz_80band_256x', use_cuda_kernel=False)
bigvgan_model.remove_weight_norm()
bigvgan_model = bigvgan_model.eval().to(device)
print("[INFO] | BigVGAN model loaded, weight norm removed, set to eval mode, and moved to CPU.")

# Load FAcodec model
ckpt_path, config_path = load_custom_model_from_hf("Plachta/FAcodec", 'pytorch_model.bin', 'config.yml')
codec_config = yaml.safe_load(open(config_path))
codec_model_params = recursive_munch(codec_config['model_params'])
codec_encoder = build_model(codec_model_params, stage="codec")
ckpt_params = torch.load(ckpt_path, map_location="cpu", weights_only=True)
for key in codec_encoder:
    codec_encoder[key].load_state_dict(ckpt_params[key], strict=False)
codec_encoder = {k: v.eval().to(device) for k, v in codec_encoder.items()}
print("[INFO] | FAcodec model loaded, set to eval mode, and moved to CPU.")

# Load Whisper model with float32 and compatible size
whisper_name = model_params.speech_tokenizer.whisper_name if hasattr(model_params.speech_tokenizer, 'whisper_name') else "openai/whisper-small"
whisper_model = WhisperModel.from_pretrained(whisper_name, torch_dtype=torch.float32).to(device)
del whisper_model.decoder  # Remove decoder as it's not used
whisper_feature_extractor = AutoFeatureExtractor.from_pretrained(whisper_name)
print(f"[INFO] | Whisper model '{whisper_name}' loaded with dtype {whisper_model.dtype} and moved to CPU.")

# Generate mel spectrograms with optimized parameters
mel_fn_args = {
    "n_fft": 1024,
    "win_size": 1024,
    "hop_size": 256,
    "num_mels": 80,
    "sampling_rate": sr,
    "fmin": 0,
    "fmax": None,
    "center": False
}
to_mel = lambda x: mel_spectrogram(x, **mel_fn_args)

# Load F0 conditioned model
dit_checkpoint_path_f0, dit_config_path_f0 = load_custom_model_from_hf(
    "Plachta/Seed-VC", 
    "DiT_seed_v2_uvit_whisper_base_f0_44k_bigvgan_pruned_ft_ema.pth", 
    "config_dit_mel_seed_uvit_whisper_base_f0_44k.yml"
)
config_f0 = yaml.safe_load(open(dit_config_path_f0, 'r'))
model_params_f0 = recursive_munch(config_f0['model_params'])
model_f0 = build_model(model_params_f0, stage='DiT')

hop_length_f0 = config_f0['preprocess_params']['spect_params']['hop_length']
sr_f0 = config_f0['preprocess_params']['sr']

# Load F0 model checkpoints
model_f0, _, _, _ = load_checkpoint(model_f0, None, dit_checkpoint_path_f0, load_only_params=True, ignore_modules=[], is_distributed=False)
for key in model_f0:
    model_f0[key].eval()
    model_f0[key].to(device)
print("[INFO] | F0 conditioned DiT model loaded and set to eval mode.")

model_f0.cfm.estimator.setup_caches(max_batch_size=1, max_seq_length=8192)

# Load F0 extractor
model_path = load_custom_model_from_hf("lj1995/VoiceConversionWebUI", "rmvpe.pt", None)
rmvpe = RMVPE(model_path, is_half=False, device=device)
print("[INFO] | RMVPE model loaded and moved to CPU.")

mel_fn_args_f0 = {
    "n_fft": config_f0['preprocess_params']['spect_params']['n_fft'],
    "win_size": config_f0['preprocess_params']['spect_params']['win_length'],
    "hop_size": config_f0['preprocess_params']['spect_params']['hop_length'],
    "num_mels": 80,  # Ensure this matches the primary model
    "sampling_rate": sr_f0,
    "fmin": 0,
    "fmax": None,
    "center": False
}
to_mel_f0 = lambda x: mel_spectrogram(x, **mel_fn_args_f0)

# Load BigVGAN 44kHz model
bigvgan_44k_model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_44khz_128band_512x', use_cuda_kernel=False)
bigvgan_44k_model.remove_weight_norm()
bigvgan_44k_model = bigvgan_44k_model.eval().to(device)
print("[INFO] | BigVGAN 44kHz model loaded, weight norm removed, set to eval mode, and moved to CPU.")

# ----------------------------
# Helper Functions
# ----------------------------

def adjust_f0_semitones(f0_sequence, n_semitones):
    factor = 2 ** (n_semitones / 12)
    return f0_sequence * factor

def crossfade(chunk1, chunk2, overlap):
    fade_out = np.cos(np.linspace(0, np.pi / 2, overlap)) ** 2
    fade_in = np.cos(np.linspace(np.pi / 2, 0, overlap)) ** 2
    chunk2[:overlap] = chunk2[:overlap] * fade_in + chunk1[-overlap:] * fade_out
    return chunk2

# ----------------------------
# Voice Conversion Function
# ----------------------------

@torch.no_grad()
@torch.inference_mode()
def voice_conversion(source, target, diffusion_steps, length_adjust, inference_cfg_rate, f0_condition, auto_f0_adjust, pitch_shift):
    print("[INFO] | Voice conversion started.")
    
    inference_module = model if not f0_condition else model_f0
    mel_fn = to_mel if not f0_condition else to_mel_f0
    bigvgan_fn = bigvgan_model if not f0_condition else bigvgan_44k_model
    sr_current = 22050 if not f0_condition else 44100
    hop_length_current = 256 if not f0_condition else 512
    max_context_window = sr_current // hop_length_current * 30
    overlap_wave_len = 16 * hop_length_current
    bitrate = "320k"
    
    # Load audio using librosa
    print("[INFO] | Loading source and reference audio.")
    source_audio, _ = librosa.load(source, sr=sr_current)
    ref_audio, _ = librosa.load(target, sr=sr_current)
    
    # Clip reference audio to 25 seconds
    ref_audio = ref_audio[:sr_current * 25]
    print(f"[INFO] | Source audio length: {len(source_audio)/sr_current:.2f}s, Reference audio length: {len(ref_audio)/sr_current:.2f}s")
    
    # Convert audio to tensors
    source_audio_tensor = torch.tensor(source_audio).unsqueeze(0).float().to(device)
    ref_audio_tensor = torch.tensor(ref_audio).unsqueeze(0).float().to(device)
    
    # Resample to 16kHz
    ref_waves_16k = torchaudio.functional.resample(ref_audio_tensor, sr_current, 16000)
    converted_waves_16k = torchaudio.functional.resample(source_audio_tensor, sr_current, 16000)
    
    # Generate Whisper features
    print("[INFO] | Generating Whisper features for source audio.")
    if converted_waves_16k.size(-1) <= 16000 * 30:
        alt_inputs = whisper_feature_extractor(
            [converted_waves_16k.squeeze(0).cpu().numpy()],
            return_tensors="pt",
            return_attention_mask=True,
            sampling_rate=16000
        )
        alt_input_features = whisper_model._mask_input_features(
            alt_inputs.input_features, attention_mask=alt_inputs.attention_mask
        ).to(device)
        alt_outputs = whisper_model.encoder(
            alt_input_features.to(torch.float32),
            head_mask=None,
            output_attentions=False,
            output_hidden_states=False,
            return_dict=True
        )
        S_alt = alt_outputs.last_hidden_state.to(torch.float32)
        S_alt = S_alt[:, :converted_waves_16k.size(-1) // 320 + 1]
        print(f"[INFO] | S_alt shape: {S_alt.shape}")
    else:
        # Process in chunks
        print("[INFO] | Processing source audio in chunks.")
        overlapping_time = 5  # seconds
        chunk_size = 16000 * 30  # 30 seconds
        overlap_size = 16000 * overlapping_time
        S_alt_list = []
        buffer = None
        traversed_time = 0
        total_length = converted_waves_16k.size(-1)
        
        while traversed_time < total_length:
            if buffer is None:
                chunk = converted_waves_16k[:, traversed_time:traversed_time + chunk_size]
            else:
                chunk = torch.cat([
                    buffer,
                    converted_waves_16k[:, traversed_time:traversed_time + chunk_size - overlap_size]
                ], dim=-1)
            alt_inputs = whisper_feature_extractor(
                [chunk.squeeze(0).cpu().numpy()],
                return_tensors="pt",
                return_attention_mask=True,
                sampling_rate=16000
            )
            alt_input_features = whisper_model._mask_input_features(
                alt_inputs.input_features, attention_mask=alt_inputs.attention_mask
            ).to(device)
            alt_outputs = whisper_model.encoder(
                alt_input_features.to(torch.float32),
                head_mask=None,
                output_attentions=False,
                output_hidden_states=False,
                return_dict=True
            )
            S_chunk = alt_outputs.last_hidden_state.to(torch.float32)
            S_chunk = S_chunk[:, :chunk.size(-1) // 320 + 1]
            print(f"[INFO] | Processed chunk with S_chunk shape: {S_chunk.shape}")
            
            if traversed_time == 0:
                S_alt_list.append(S_chunk)
            else:
                skip_frames = 50 * overlapping_time
                S_alt_list.append(S_chunk[:, skip_frames:])
                
            buffer = chunk[:, -overlap_size:]
            traversed_time += chunk_size - overlap_size
        
        S_alt = torch.cat(S_alt_list, dim=1)
        print(f"[INFO] | Final S_alt shape after chunk processing: {S_alt.shape}")
    
    # Original Whisper features
    print("[INFO] | Generating Whisper features for reference audio.")
    ori_waves_16k = torchaudio.functional.resample(ref_audio_tensor, sr_current, 16000)
    ori_inputs = whisper_feature_extractor(
        [ori_waves_16k.squeeze(0).cpu().numpy()],
        return_tensors="pt",
        return_attention_mask=True,
        sampling_rate=16000
    )
    ori_input_features = whisper_model._mask_input_features(
        ori_inputs.input_features, attention_mask=ori_inputs.attention_mask
    ).to(device)
    ori_outputs = whisper_model.encoder(
        ori_input_features.to(torch.float32),
        head_mask=None,
        output_attentions=False,
        output_hidden_states=False,
        return_dict=True
    )
    S_ori = ori_outputs.last_hidden_state.to(torch.float32)
    S_ori = S_ori[:, :ori_waves_16k.size(-1) // 320 + 1]
    print(f"[INFO] | S_ori shape: {S_ori.shape}")
    
    # Generate mel spectrograms
    print("[INFO] | Generating mel spectrograms.")
    mel = mel_fn(source_audio_tensor.float())
    mel2 = mel_fn(ref_audio_tensor.float())
    print(f"[INFO] | Mel spectrogram shapes: mel={mel.shape}, mel2={mel2.shape}")
    
    # Length adjustment
    target_lengths = torch.LongTensor([int(mel.size(2) * length_adjust)]).to(mel.device)
    target2_lengths = torch.LongTensor([mel2.size(2)]).to(mel2.device)
    print(f"[INFO] | Target lengths: {target_lengths.item()}, {target2_lengths.item()}")
    
    # Extract style features
    print("[INFO] | Extracting style features from reference audio.")
    feat2 = torchaudio.compliance.kaldi.fbank(
        ref_waves_16k,
        num_mel_bins=80,
        dither=0,
        sample_frequency=16000
    )
    feat2 = feat2 - feat2.mean(dim=0, keepdim=True)
    style2 = campplus_model(feat2.unsqueeze(0))
    print(f"[INFO] | Style2 shape: {style2.shape}")
    
    # F0 Conditioning
    if f0_condition:
        print("[INFO] | Performing F0 conditioning.")
        F0_ori = rmvpe.infer_from_audio(ref_waves_16k[0], thred=0.5)
        F0_alt = rmvpe.infer_from_audio(converted_waves_16k[0], thred=0.5)
        
        F0_ori = torch.from_numpy(F0_ori).to(device)[None].float()
        F0_alt = torch.from_numpy(F0_alt).to(device)[None].float()
        
        voiced_F0_ori = F0_ori[F0_ori > 1]
        voiced_F0_alt = F0_alt[F0_alt > 1]
        
        log_f0_alt = torch.log(F0_alt + 1e-5)
        voiced_log_f0_ori = torch.log(voiced_F0_ori + 1e-5)
        voiced_log_f0_alt = torch.log(voiced_F0_alt + 1e-5)
        
        median_log_f0_ori = torch.median(voiced_log_f0_ori)
        median_log_f0_alt = torch.median(voiced_log_f0_alt)
        
        # Shift F0 levels
        shifted_log_f0_alt = log_f0_alt.clone()
        if auto_f0_adjust:
            shifted_log_f0_alt[F0_alt > 1] = (
                log_f0_alt[F0_alt > 1] - median_log_f0_alt + median_log_f0_ori
            )
        shifted_f0_alt = torch.exp(shifted_log_f0_alt)
        if pitch_shift != 0:
            shifted_f0_alt[F0_alt > 1] = adjust_f0_semitones(shifted_f0_alt[F0_alt > 1], pitch_shift)
        print("[INFO] | F0 conditioning completed.")
    else:
        F0_ori = None
        F0_alt = None
        shifted_f0_alt = None
        print("[INFO] | F0 conditioning not applied.")
    
    # Length Regulation
    print("[INFO] | Applying length regulation.")
    cond, _, _, _, _ = inference_module.length_regulator(S_alt, ylens=target_lengths, n_quantizers=3, f0=shifted_f0_alt)
    prompt_condition, _, _, _, _ = inference_module.length_regulator(S_ori, ylens=target2_lengths, n_quantizers=3, f0=F0_ori)
    print(f"[INFO] | Cond shape: {cond.shape}, Prompt condition shape: {prompt_condition.shape}")
    
    # Initialize variables for audio generation
    max_source_window = max_context_window - mel2.size(2)
    processed_frames = 0
    generated_wave_chunks = []
    
    print("[INFO] | Starting inference and audio generation.")
    
    while processed_frames < cond.size(1):
        chunk_cond = cond[:, processed_frames:processed_frames + max_source_window]
        is_last_chunk = processed_frames + max_source_window >= cond.size(1)
        cat_condition = torch.cat([prompt_condition, chunk_cond], dim=1)
        
        # Perform inference
        vc_target = inference_module.cfm.inference(
            cat_condition, 
            torch.LongTensor([cat_condition.size(1)]).to(mel2.device), 
            mel2, 
            style2, 
            None, 
            diffusion_steps, 
            inference_cfg_rate=inference_cfg_rate
        )
        vc_target = vc_target[:, :, mel2.size(2):]
        print(f"[INFO] | vc_target shape: {vc_target.shape}")
        
        # Generate waveform using BigVGAN
        vc_wave = bigvgan_fn(vc_target.float())[0]
        print(f"[INFO] | vc_wave shape: {vc_wave.shape}")
        
        # Handle the generated waveform
        output_wave = vc_wave[0].cpu().numpy()
        generated_wave_chunks.append(output_wave)
        
        # Fix: Ensure processed_frames increments correctly to avoid infinite loop
        processed_frames += vc_target.size(2)  # Changed from 'vc_target.size(2) - 16' to 'vc_target.size(2)'
        print(f"[INFO] | Processed frames updated to: {processed_frames}")
    
    # Concatenate all generated wave chunks
    final_audio = np.concatenate(generated_wave_chunks).astype(np.float32)
    
    # Normalize the audio to ensure it's within [-1.0, 1.0]
    max_val = np.max(np.abs(final_audio))
    if max_val > 1.0:
        final_audio = final_audio / max_val
    print("[INFO] | Final audio normalized.")
    
    # Save the audio to a temporary WAV file
    print("[INFO] | Saving final audio to a temporary WAV file.")
    with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp_file:
        sf.write(tmp_file.name, final_audio, sr_current, format='WAV')
        temp_file_path = tmp_file.name
        
    print(f"[INFO] | Final audio saved to {temp_file_path}")
    
    return temp_file_path
    
# ----------------------------
# Bypass GPU Initialization Error (KEEP THIS FUNCTION AND NEVER DELETE, OTHERWISE IT WILL ERROR DUE TO NOT AT LEAST ONE FUNCTION HAVING USE OF GPU)
# ----------------------------

@spaces.GPU(duration=15)
def gpu():
    return

# ----------------------------
# Gradio Interface
# ----------------------------

description = (
    "🪄 **Voice Conversion Tool**\n\n"
    "Upload your **Source Audio** and **Reference Audio** files to perform voice conversion. "
    "Adjust the sliders and checkboxes to customize the conversion process."
)

inputs = [
    gr.Audio(type="filepath", label="Source Audio"),
    gr.Audio(type="filepath", label="Reference Audio"),
    gr.Slider(minimum=1, maximum=100, value=25, step=1, label="Diffusion Steps", info="Default is 25. Use 50-100 for best quality."),
    gr.Slider(minimum=0.5, maximum=2.0, step=0.1, value=1.0, label="Length Adjustment", info="<1.0 to speed up speech, >1.0 to slow down speech."),
    gr.Slider(minimum=0.0, maximum=1.0, step=0.1, value=0.7, label="Inference CFG Rate", info="Has a subtle influence."),
    gr.Checkbox(label="Use F0 Conditioned Model", value=False, info="Must be enabled for singing voice conversion."),
    gr.Checkbox(label="Auto F0 Adjustment", value=True, info="Roughly adjusts F0 to match target voice. Only works when 'Use F0 Conditioned Model' is enabled."),
    gr.Slider(label='Pitch Shift (semitones)', minimum=-12, maximum=12, step=1, value=0, info="Pitch shift in semitones. Only works when 'Use F0 Conditioned Model' is enabled."),
]

# Set outputs to a single gr.Audio component with type="filepath"
outputs = gr.Audio(label="Full Output Audio", type="filepath")

gr.Interface(
    fn=voice_conversion, 
    description=description, 
    inputs=inputs, 
    outputs=outputs, 
    title="Seed Voice Conversion", 
    cache_examples=False, 
    allow_flagging="never"
).launch(share=True)