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Zero
| import spaces | |
| 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 | |
| # Load model and configuration | |
| 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_seed_v2_uvit_whisper_small_wavenet_bigvgan_pruned.pth", | |
| "config_dit_mel_seed_uvit_whisper_small_wavenet.yml") | |
| # dit_checkpoint_path = "E:/DiT_epoch_00018_step_801000.pth" | |
| # dit_config_path = "configs/config_dit_mel_seed_uvit_whisper_small_encoder_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'] | |
| # Load 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) | |
| model.cfm.estimator.setup_caches(max_batch_size=1, max_seq_length=8192) | |
| # Load additional modules | |
| 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.bigvgan import bigvgan | |
| bigvgan_model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_22khz_80band_256x', use_cuda_kernel=False) | |
| # remove weight norm in the model and set to eval mode | |
| bigvgan_model.remove_weight_norm() | |
| bigvgan_model = bigvgan_model.eval().to(device) | |
| 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] | |
| # whisper | |
| from transformers import AutoFeatureExtractor, WhisperModel | |
| 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.float16).to(device) | |
| del whisper_model.decoder | |
| whisper_feature_extractor = AutoFeatureExtractor.from_pretrained(whisper_name) | |
| # Generate mel spectrograms | |
| 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": None, | |
| "center": False | |
| } | |
| from modules.audio import mel_spectrogram | |
| to_mel = lambda x: mel_spectrogram(x, **mel_fn_args) | |
| # f0 conditioned model | |
| dit_checkpoint_path, dit_config_path = 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 = 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'] | |
| # Load checkpoints | |
| 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 extractor | |
| 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) | |
| 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 | |
| } | |
| to_mel_f0 = lambda x: mel_spectrogram(x, **mel_fn_args_f0) | |
| bigvgan_44k_model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_44khz_128band_512x', use_cuda_kernel=False) | |
| # remove weight norm in the model and set to eval mode | |
| bigvgan_44k_model.remove_weight_norm() | |
| bigvgan_44k_model = bigvgan_44k_model.eval().to(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.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 | |
| # streaming and chunk processing related params | |
| bitrate = "320k" | |
| overlap_frame_len = 16 | |
| def voice_conversion(source, target, diffusion_steps, length_adjust, inference_cfg_rate, 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 | |
| bigvgan_fn = bigvgan_model if not f0_condition else bigvgan_44k_model | |
| sr = 22050 if not f0_condition else 44100 | |
| hop_length = 256 if not f0_condition else 512 | |
| max_context_window = sr // hop_length * 30 | |
| overlap_wave_len = overlap_frame_len * hop_length | |
| # Load audio | |
| source_audio = librosa.load(source, sr=sr)[0] | |
| ref_audio = librosa.load(target, sr=sr)[0] | |
| # Process audio | |
| source_audio = torch.tensor(source_audio).unsqueeze(0).float().to(device) | |
| ref_audio = torch.tensor(ref_audio[:sr * 25]).unsqueeze(0).float().to(device) | |
| # Resample | |
| ref_waves_16k = torchaudio.functional.resample(ref_audio, sr, 16000) | |
| converted_waves_16k = torchaudio.functional.resample(source_audio, sr, 16000) | |
| # if source audio less than 30 seconds, whisper can handle in one forward | |
| 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(whisper_model.encoder.dtype), | |
| 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] | |
| else: | |
| overlapping_time = 5 # 5 seconds | |
| S_alt_list = [] | |
| buffer = None | |
| traversed_time = 0 | |
| while traversed_time < converted_waves_16k.size(-1): | |
| if buffer is None: # first chunk | |
| chunk = converted_waves_16k[:, traversed_time:traversed_time + 16000 * 30] | |
| else: | |
| chunk = torch.cat([buffer, converted_waves_16k[:, traversed_time:traversed_time + 16000 * (30 - overlapping_time)]], 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(whisper_model.encoder.dtype), | |
| 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[:, :chunk.size(-1) // 320 + 1] | |
| if traversed_time == 0: | |
| S_alt_list.append(S_alt) | |
| else: | |
| S_alt_list.append(S_alt[:, 50 * overlapping_time:]) | |
| buffer = chunk[:, -16000 * overlapping_time:] | |
| traversed_time += 30 * 16000 if traversed_time == 0 else chunk.size(-1) - 16000 * overlapping_time | |
| S_alt = torch.cat(S_alt_list, dim=1) | |
| ori_waves_16k = torchaudio.functional.resample(ref_audio, sr, 16000) | |
| ori_inputs = whisper_feature_extractor([ori_waves_16k.squeeze(0).cpu().numpy()], | |
| return_tensors="pt", | |
| return_attention_mask=True) | |
| ori_input_features = whisper_model._mask_input_features( | |
| ori_inputs.input_features, attention_mask=ori_inputs.attention_mask).to(device) | |
| with torch.no_grad(): | |
| ori_outputs = whisper_model.encoder( | |
| ori_input_features.to(whisper_model.encoder.dtype), | |
| 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] | |
| 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: | |
| 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] | |
| 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) | |
| # shift alt log f0 level to ori log f0 level | |
| 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 | |
| # Length regulation | |
| cond, _, codes, commitment_loss, codebook_loss = inference_module.length_regulator(S_alt, ylens=target_lengths, n_quantizers=3, f0=shifted_f0_alt) | |
| prompt_condition, _, codes, commitment_loss, codebook_loss = inference_module.length_regulator(S_ori, ylens=target2_lengths, n_quantizers=3, f0=F0_ori) | |
| max_source_window = max_context_window - mel2.size(2) | |
| # split source condition (cond) into chunks | |
| processed_frames = 0 | |
| generated_wave_chunks = [] | |
| # generate chunk by chunk and stream the output | |
| 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) | |
| with torch.autocast(device_type='cuda', dtype=torch.float16): | |
| # Voice Conversion | |
| 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):] | |
| vc_wave = bigvgan_fn(vc_target.float())[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 = ("State-of-the-Art zero-shot voice conversion/singing voice conversion. For local deployment please check [GitHub repository](https://github.com/Plachtaa/seed-vc) " | |
| "for details and updates.<br>Note that any reference audio will be forcefully clipped to 25s if beyond this length.<br> " | |
| "If total duration of source and reference audio exceeds 30s, source audio will be processed in chunks.<br> " | |
| "无需训练的 zero-shot 语音/歌声转换模型，若需本地部署查看[GitHub页面](https://github.com/Plachtaa/seed-vc)<br>" | |
| "请注意，参考音频若超过 25 秒，则会被自动裁剪至此长度。<br>若源音频和参考音频的总时长超过 30 秒，源音频将被分段处理。") | |
| inputs = [ | |
| gr.Audio(type="filepath", label="Source Audio / 源音频"), | |
| gr.Audio(type="filepath", label="Reference Audio / 参考音频"), | |
| gr.Slider(minimum=1, maximum=200, value=25, step=1, label="Diffusion Steps / 扩散步数", info="25 by default, 50~100 for best quality / 默认为 25，50~100 为最佳质量"), | |
| gr.Slider(minimum=0.5, maximum=2.0, step=0.1, value=1.0, label="Length Adjust / 长度调整", info="<1.0 for speed-up speech, >1.0 for slow-down speech / <1.0 加速语速，>1.0 减慢语速"), | |
| gr.Slider(minimum=0.0, maximum=1.0, step=0.1, value=0.7, label="Inference CFG Rate", info="has subtle influence / 有微小影响"), | |
| gr.Checkbox(label="Use F0 conditioned model / 启用F0输入", value=False, info="Must set to true for singing voice conversion / 歌声转换时必须勾选"), | |
| gr.Checkbox(label="Auto F0 adjust / 自动F0调整", value=True, | |
| info="Roughly adjust F0 to match target voice. Only works when F0 conditioned model is used. / 粗略调整 F0 以匹配目标音色，仅在勾选 '启用F0输入' 时生效"), | |
| gr.Slider(label='Pitch shift / 音调变换', minimum=-24, maximum=24, step=1, value=0, info="Pitch shift in semitones, only works when F0 conditioned model is used / 半音数的音高变换，仅在勾选 '启用F0输入' 时生效"), | |
| ] | |
| examples = [["examples/source/yae_0.wav", "examples/reference/dingzhen_0.wav", 25, 1.0, 0.7, False, True, 0], | |
| ["examples/source/jay_0.wav", "examples/reference/azuma_0.wav", 25, 1.0, 0.7, False, True, 0], | |
| ["examples/source/Wiz Khalifa,Charlie Puth - See You Again [vocals]_[cut_28sec].wav", | |
| "examples/reference/kobe_0.wav", 50, 1.0, 0.7, True, False, -6], | |
| ["examples/source/TECHNOPOLIS - 2085 [vocals]_[cut_14sec].wav", | |
| "examples/reference/trump_0.wav", 50, 1.0, 0.7, True, False, -12], | |
| ] | |
| outputs = [gr.Audio(label="Stream Output Audio / 流式输出", streaming=True, format='mp3'), | |
| gr.Audio(label="Full Output Audio / 完整输出", streaming=False, format='wav')] | |
| gr.Interface(fn=voice_conversion, | |
| description=description, | |
| inputs=inputs, | |
| outputs=outputs, | |
| title="Seed Voice Conversion", | |
| examples=examples, | |
| cache_examples=False, | |
| ).launch() |