app.py

import gradio as gr
import torch
import torchaudio
import librosa
from modules.commons import build_model, load_checkpoint, recursive_munch
import yaml
from hf_utils import load_custom_model_from_hf
import numpy as np
from pydub import AudioSegment
import spaces

# 모델 및 설정 로드
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")

dit_checkpoint_path, dit_config_path = load_custom_model_from_hf("Plachta/Seed-VC",
                                                "DiT_step_298000_seed_uvit_facodec_small_wavenet_pruned.pth",
                                                "config_dit_mel_seed_facodec_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')
hop_length = config['preprocess_params']['spect_params']['hop_length']
sr = config['preprocess_params']['sr']

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

# 추가 모듈 로드
from modules.campplus.DTDNN import CAMPPlus

campplus_ckpt_path = load_custom_model_from_hf("funasr/campplus", "campplus_cn_common.bin", config_filename=None)
campplus_model = CAMPPlus(feat_dim=80, embedding_size=192)
campplus_model.load_state_dict(torch.load(campplus_ckpt_path, map_location="cpu"))
campplus_model.eval()
campplus_model.to(device)

from modules.hifigan.generator import HiFTGenerator
from modules.hifigan.f0_predictor import ConvRNNF0Predictor

hift_checkpoint_path, hift_config_path = load_custom_model_from_hf("Plachta/Seed-VC",
                                                "hift.pt",
                                                "hifigan.yml")
hift_config = yaml.safe_load(open(hift_config_path, 'r'))
hift_gen = HiFTGenerator(**hift_config['hift'], f0_predictor=ConvRNNF0Predictor(**hift_config['f0_predictor']))
hift_gen.load_state_dict(torch.load(hift_checkpoint_path, map_location='cpu'))
hift_gen.eval()
hift_gen.to(device)

from modules.bigvgan import bigvgan

bigvgan_model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_22khz_80band_256x', use_cuda_kernel=False)

# 모델에서 weight norm을 제거하고 eval 모드로 설정
bigvgan_model.remove_weight_norm()
bigvgan_model = bigvgan_model.eval().to(device)

speech_tokenizer_type = config['model_params']['speech_tokenizer'].get('type', 'cosyvoice')
if speech_tokenizer_type == 'cosyvoice':
    from modules.cosyvoice_tokenizer.frontend import CosyVoiceFrontEnd
    speech_tokenizer_path = load_custom_model_from_hf("Plachta/Seed-VC", "speech_tokenizer_v1.onnx", None)
    cosyvoice_frontend = CosyVoiceFrontEnd(speech_tokenizer_model=speech_tokenizer_path,
                                           device='cuda', device_id=0)
elif speech_tokenizer_type == 'facodec':
    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")

    for key in codec_encoder:
        codec_encoder[key].load_state_dict(ckpt_params[key], strict=False)
    _ = [codec_encoder[key].eval() for key in codec_encoder]
    _ = [codec_encoder[key].to(device) for key in codec_encoder]

# mel 스펙트로그램 생성
mel_fn_args = {
    "n_fft": config['preprocess_params']['spect_params']['n_fft'],
    "win_size": config['preprocess_params']['spect_params']['win_length'],
    "hop_size": config['preprocess_params']['spect_params']['hop_length'],
    "num_mels": config['preprocess_params']['spect_params']['n_mels'],
    "sampling_rate": sr,
    "fmin": 0,
    "fmax": 8000,
    "center": False
}
mel_fn_args_f0 = {
    "n_fft": config['preprocess_params']['spect_params']['n_fft'],
    "win_size": config['preprocess_params']['spect_params']['win_length'],
    "hop_size": config['preprocess_params']['spect_params']['hop_length'],
    "num_mels": config['preprocess_params']['spect_params']['n_mels'],
    "sampling_rate": sr,
    "fmin": 0,
    "fmax": None,
    "center": False
}
from modules.audio import mel_spectrogram

to_mel = lambda x: mel_spectrogram(x, **mel_fn_args)
to_mel_f0 = lambda x: mel_spectrogram(x, **mel_fn_args_f0)

# f0 조건부 모델
dit_checkpoint_path, dit_config_path = load_custom_model_from_hf("Plachta/Seed-VC",
                                                "DiT_seed_v2_uvit_facodec_small_wavenet_f0_bigvgan_pruned.pth",
                                                "config_dit_mel_seed_facodec_small_wavenet_f0.yml")

config = yaml.safe_load(open(dit_config_path, 'r'))
model_params = recursive_munch(config['model_params'])
model_f0 = build_model(model_params, stage='DiT')
hop_length = config['preprocess_params']['spect_params']['hop_length']
sr = config['preprocess_params']['sr']

# 체크포인트 로드
model_f0, _, _, _ = load_checkpoint(model_f0, None, dit_checkpoint_path,
                                 load_only_params=True, ignore_modules=[], is_distributed=False)
for key in model_f0:
    model_f0[key].eval()
    model_f0[key].to(device)
model_f0.cfm.estimator.setup_caches(max_batch_size=1, max_seq_length=8192)

# f0 추출기
from modules.rmvpe import RMVPE

model_path = load_custom_model_from_hf("lj1995/VoiceConversionWebUI", "rmvpe.pt", None)
rmvpe = RMVPE(model_path, is_half=False, device=device)

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.linspace(1, 0, overlap)
    fade_in = np.linspace(0, 1, overlap)
    chunk2[:overlap] = chunk2[:overlap] * fade_in + chunk1[-overlap:] * fade_out
    return chunk2

# 스트리밍 및 청크 처리 관련 매개변수
max_context_window = sr // hop_length * 30
overlap_frame_len = 64
overlap_wave_len = overlap_frame_len * hop_length
bitrate = "320k"

@spaces.GPU
@torch.no_grad()
@torch.inference_mode()
def voice_conversion(source, target, diffusion_steps, length_adjust, inference_cfg_rate, n_quantizers, f0_condition, auto_f0_adjust, pitch_shift):
    inference_module = model if not f0_condition else model_f0
    mel_fn = to_mel if not f0_condition else to_mel_f0
    # 오디오 로드
    source_audio = librosa.load(source, sr=sr)[0]
    ref_audio = librosa.load(target, sr=sr)[0]

    # 오디오 처리
    source_audio = torch.tensor(source_audio).unsqueeze(0).float().to(device)
    ref_audio = torch.tensor(ref_audio[:sr * 25]).unsqueeze(0).float().to(device)

    # 리샘플링
    source_waves_16k = torchaudio.functional.resample(source_audio, sr, 16000)
    ref_waves_16k = torchaudio.functional.resample(ref_audio, sr, 16000)

    # 특성 추출
    if speech_tokenizer_type == 'cosyvoice':
        S_alt = cosyvoice_frontend.extract_speech_token(source_waves_16k)[0]
        S_ori = cosyvoice_frontend.extract_speech_token(ref_waves_16k)[0]
    elif speech_tokenizer_type == 'facodec':
        converted_waves_24k = torchaudio.functional.resample(source_audio, sr, 24000)
        waves_input = converted_waves_24k.unsqueeze(1)
        max_wave_len_per_chunk = 24000 * 20
        wave_input_chunks = [
            waves_input[..., i:i + max_wave_len_per_chunk] for i in range(0, waves_input.size(-1), max_wave_len_per_chunk)
        ]
        S_alt_chunks = []
        for i, chunk in enumerate(wave_input_chunks):
            z = codec_encoder.encoder(chunk)
            (
                quantized,
                codes
            ) = codec_encoder.quantizer(
                z,
                chunk,
            )
            S_alt = torch.cat([codes[1], codes[0]], dim=1)
            S_alt_chunks.append(S_alt)
        S_alt = torch.cat(S_alt_chunks, dim=-1)

        # S_ori도 동일한 방식으로 추출해야 함
        waves_24k = torchaudio.functional.resample(ref_audio, sr, 24000)
        waves_input = waves_24k.unsqueeze(1)
        z = codec_encoder.encoder(waves_input)
        (
            quantized,
            codes
        ) = codec_encoder.quantizer(
            z,
            waves_input,
        )
        S_ori = torch.cat([codes[1], codes[0]], dim=1)

    mel = mel_fn(source_audio.to(device).float())
    mel2 = mel_fn(ref_audio.to(device).float())

    target_lengths = torch.LongTensor([int(mel.size(2) * length_adjust)]).to(mel.device)
    target2_lengths = torch.LongTensor([mel2.size(2)]).to(mel2.device)

    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))

    if f0_condition:
        waves_16k = torchaudio.functional.resample(waves_24k, sr, 16000)
        converted_waves_16k = torchaudio.functional.resample(converted_waves_24k, sr, 16000)
        F0_ori = rmvpe.infer_from_audio(waves_16k[0], thred=0.03)
        F0_alt = rmvpe.infer_from_audio(converted_waves_16k[0], thred=0.03)

        F0_ori = torch.from_numpy(F0_ori).to(device)[None]
        F0_alt = torch.from_numpy(F0_alt).to(device)[None]

        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)
        # mean_log_f0_ori = torch.mean(voiced_log_f0_ori)
        # mean_log_f0_alt = torch.mean(voiced_log_f0_alt)

        # alt log f0 레벨을 ori log f0 레벨로 이동
        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)
    else:
        F0_ori = None
        F0_alt = None
        shifted_f0_alt = None

    # 길이 조절
    cond = inference_module.length_regulator(S_alt, ylens=target_lengths, n_quantizers=int(n_quantizers), f0=shifted_f0_alt)[0]
    prompt_condition = inference_module.length_regulator(S_ori, ylens=target2_lengths, n_quantizers=int(n_quantizers), f0=F0_ori)[0]

    max_source_window = max_context_window - mel2.size(2)
    # 소스 조건(cond)을 청크로 분할
    processed_frames = 0
    generated_wave_chunks = []
    # 청크별로 생성하고 출력을 스트리밍
    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)
        # 음성 변환
        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(-1):]
        if not f0_condition:
            vc_wave = hift_gen.inference(vc_target, f0=None)
        else:
            vc_wave = bigvgan_model(vc_target)[0]
        if processed_frames == 0:
            if is_last_chunk:
                output_wave = vc_wave[0].cpu().numpy()
                generated_wave_chunks.append(output_wave)
                output_wave = (output_wave * 32768.0).astype(np.int16)
                mp3_bytes = AudioSegment(
                    output_wave.tobytes(), frame_rate=sr,
                    sample_width=output_wave.dtype.itemsize, channels=1
                ).export(format="mp3", bitrate=bitrate).read()
                yield mp3_bytes, (sr, np.concatenate(generated_wave_chunks))
                break
            output_wave = vc_wave[0, :-overlap_wave_len].cpu().numpy()
            generated_wave_chunks.append(output_wave)
            previous_chunk = vc_wave[0, -overlap_wave_len:]
            processed_frames += vc_target.size(2) - overlap_frame_len
            output_wave = (output_wave * 32768.0).astype(np.int16)
            mp3_bytes = AudioSegment(
                output_wave.tobytes(), frame_rate=sr,
                sample_width=output_wave.dtype.itemsize, channels=1
            ).export(format="mp3", bitrate=bitrate).read()
            yield mp3_bytes, None
        elif is_last_chunk:
            output_wave = crossfade(previous_chunk.cpu().numpy(), vc_wave[0].cpu().numpy(), overlap_wave_len)
            generated_wave_chunks.append(output_wave)
            processed_frames += vc_target.size(2) - overlap_frame_len
            output_wave = (output_wave * 32768.0).astype(np.int16)
            mp3_bytes = AudioSegment(
                output_wave.tobytes(), frame_rate=sr,
                sample_width=output_wave.dtype.itemsize, channels=1
            ).export(format="mp3", bitrate=bitrate).read()
            yield mp3_bytes, (sr, np.concatenate(generated_wave_chunks))
            break
        else:
            output_wave = crossfade(previous_chunk.cpu().numpy(), vc_wave[0, :-overlap_wave_len].cpu().numpy(), overlap_wave_len)
            generated_wave_chunks.append(output_wave)
            previous_chunk = vc_wave[0, -overlap_wave_len:]
            processed_frames += vc_target.size(2) - overlap_frame_len
            output_wave = (output_wave * 32768.0).astype(np.int16)
            mp3_bytes = AudioSegment(
                output_wave.tobytes(), frame_rate=sr,
                sample_width=output_wave.dtype.itemsize, channels=1
            ).export(format="mp3", bitrate=bitrate).read()
            yield mp3_bytes, None


if __name__ == "__main__":
    description = ("레퍼런스 음악은 25초 이내 최대 30초 미만으로 업로드 바랍니다.")

    inputs = [
        gr.Audio(type="filepath", label="음악 업로드"),
        gr.Audio(type="filepath", label="음성 업로드"),
        gr.Slider(minimum=1, maximum=200, value=10, step=1, label="확산 단계", info="기본값은 10, 최상의 품질을 위해서는 50~100"),
        gr.Slider(minimum=0.5, maximum=2.0, step=0.1, value=1.0, label="길이 조정", info="<1.0 빠른 음성, >1.0 느린 음성"),
        gr.Slider(minimum=0.0, maximum=1.0, step=0.1, value=0.7, label="추론 CFG 비율", info="미묘한 영향이 있음"),
        gr.Slider(minimum=1, maximum=3, step=1, value=3, label="FAcodec 양자화기 수", info="사용하는 FAcodec 양자화기가 적을수록 원본 오디오의 운율이 덜 보존됨"),
        gr.Checkbox(label="F0 조건부 모델 사용", value=True, info="노래 음성 변환을 위해서는 반드시 체크해야 함"),
        gr.Checkbox(label="자동 F0 조정", value=True,
                    info="목표 음색에 맞게 F0를 대략적으로 조정. F0 조건부 모델 사용 시에만 작동"),
        gr.Slider(label='음조 변경', minimum=-24, maximum=24, step=1, value=0, info="반음 단위의 음조 변경, F0 조건부 모델 사용 시에만 작동"),
    ]

    examples = [["examples/source/yae_0.wav", "examples/reference/dingzhen_0.wav", 25, 1.0, 0.7, 1, False, True, 0],
                ["examples/source/jay_0.wav", "examples/reference/azuma_0.wav", 25, 1.0, 0.7, 1, True, True, 0],
                ["examples/source/Wiz Khalifa,Charlie Puth - See You Again [vocals]_[cut_28sec].wav",
                 "examples/reference/teio_0.wav", 100, 1.0, 0.7, 1, True, False, 0],
                ["examples/source/TECHNOPOLIS - 2085 [vocals]_[cut_14sec].wav",
                 "examples/reference/trump_0.wav", 50, 1.0, 0.7, 1, True, False, -12],
                ]

    outputs = [gr.Audio(label="스트리밍 출력 오디오", streaming=True, format='mp3'),
               gr.Audio(label="전체 출력 오디오", streaming=False, format='wav')]

    gr.Interface(fn=voice_conversion,
                 description=description,
                 inputs=inputs,
                 outputs=outputs,
                 title="Seed 음성 변환",
                 examples=examples,
                 cache_examples=False,
                 ).launch()