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import spaces | |
import accelerate | |
import gradio as gr | |
import torch | |
import safetensors | |
from huggingface_hub import hf_hub_download | |
import soundfile as sf | |
import os | |
import numpy as np | |
import librosa | |
from models.codec.kmeans.repcodec_model import RepCodec | |
from models.tts.maskgct.maskgct_s2a import MaskGCT_S2A | |
from models.tts.maskgct.maskgct_t2s import MaskGCT_T2S | |
from models.codec.amphion_codec.codec import CodecEncoder, CodecDecoder | |
from transformers import Wav2Vec2BertModel | |
from utils.util import load_config | |
from models.tts.maskgct.g2p.g2p_generation import g2p, chn_eng_g2p | |
from transformers import SeamlessM4TFeatureExtractor | |
# import whisperx | |
processor = SeamlessM4TFeatureExtractor.from_pretrained("facebook/w2v-bert-2.0") | |
device = torch.device("cuda" if torch.cuda.is_available() else "CPU") | |
# whisper_model = whisperx.load_model("small", "cuda", compute_type="int8") | |
# @torch.no_grad() | |
# def get_prompt_text(speech_16k): | |
# asr_result = whisper_model.transcribe(speech_16k) | |
# print("asr_result:", asr_result) | |
# language = asr_result["language"] | |
# #text = asr_result["text"] # whisper asr result | |
# text = asr_result["segments"][0]["text"] | |
# print("prompt text:", text) | |
# return text, language | |
def g2p_(text, language): | |
if language in ["zh", "en"]: | |
return chn_eng_g2p(text) | |
else: | |
return g2p(text, sentence=None, language=language) | |
def build_t2s_model(cfg, device): | |
t2s_model = MaskGCT_T2S(cfg=cfg) | |
t2s_model.eval() | |
t2s_model.to(device) | |
return t2s_model | |
def build_s2a_model(cfg, device): | |
soundstorm_model = MaskGCT_S2A(cfg=cfg) | |
soundstorm_model.eval() | |
soundstorm_model.to(device) | |
return soundstorm_model | |
def build_semantic_model(device): | |
semantic_model = Wav2Vec2BertModel.from_pretrained("facebook/w2v-bert-2.0") | |
semantic_model.eval() | |
semantic_model.to(device) | |
stat_mean_var = torch.load("./models/tts/maskgct/ckpt/wav2vec2bert_stats.pt") | |
semantic_mean = stat_mean_var["mean"] | |
semantic_std = torch.sqrt(stat_mean_var["var"]) | |
semantic_mean = semantic_mean.to(device) | |
semantic_std = semantic_std.to(device) | |
return semantic_model, semantic_mean, semantic_std | |
def build_semantic_codec(cfg, device): | |
semantic_codec = RepCodec(cfg=cfg) | |
semantic_codec.eval() | |
semantic_codec.to(device) | |
return semantic_codec | |
def build_acoustic_codec(cfg, device): | |
codec_encoder = CodecEncoder(cfg=cfg.encoder) | |
codec_decoder = CodecDecoder(cfg=cfg.decoder) | |
codec_encoder.eval() | |
codec_decoder.eval() | |
codec_encoder.to(device) | |
codec_decoder.to(device) | |
return codec_encoder, codec_decoder | |
def extract_features(speech, processor): | |
inputs = processor(speech, sampling_rate=16000, return_tensors="pt") | |
input_features = inputs["input_features"][0] | |
attention_mask = inputs["attention_mask"][0] | |
return input_features, attention_mask | |
def extract_semantic_code(semantic_mean, semantic_std, input_features, attention_mask): | |
vq_emb = semantic_model( | |
input_features=input_features, | |
attention_mask=attention_mask, | |
output_hidden_states=True, | |
) | |
feat = vq_emb.hidden_states[17] # (B, T, C) | |
feat = (feat - semantic_mean.to(feat)) / semantic_std.to(feat) | |
semantic_code, rec_feat = semantic_codec.quantize(feat) # (B, T) | |
return semantic_code, rec_feat | |
def extract_acoustic_code(speech): | |
vq_emb = codec_encoder(speech.unsqueeze(1)) | |
_, vq, _, _, _ = codec_decoder.quantizer(vq_emb) | |
acoustic_code = vq.permute(1, 2, 0) | |
return acoustic_code | |
def text2semantic( | |
device, | |
prompt_speech, | |
prompt_text, | |
prompt_language, | |
target_text, | |
target_language, | |
target_len=None, | |
n_timesteps=50, | |
cfg=2.5, | |
rescale_cfg=0.75, | |
): | |
prompt_phone_id = g2p_(prompt_text, prompt_language)[1] | |
target_phone_id = g2p_(target_text, target_language)[1] | |
if target_len < 0: | |
target_len = int( | |
(len(prompt_speech) * len(target_phone_id) / len(prompt_phone_id)) | |
/ 16000 | |
* 50 | |
) | |
else: | |
target_len = int(target_len * 50) | |
prompt_phone_id = torch.tensor(prompt_phone_id, dtype=torch.long).to(device) | |
target_phone_id = torch.tensor(target_phone_id, dtype=torch.long).to(device) | |
phone_id = torch.cat([prompt_phone_id, target_phone_id]) | |
input_fetures, attention_mask = extract_features(prompt_speech, processor) | |
input_fetures = input_fetures.unsqueeze(0).to(device) | |
attention_mask = attention_mask.unsqueeze(0).to(device) | |
semantic_code, _ = extract_semantic_code( | |
semantic_mean, semantic_std, input_fetures, attention_mask | |
) | |
predict_semantic = t2s_model.reverse_diffusion( | |
semantic_code[:, :], | |
target_len, | |
phone_id.unsqueeze(0), | |
n_timesteps=n_timesteps, | |
cfg=cfg, | |
rescale_cfg=rescale_cfg, | |
) | |
combine_semantic_code = torch.cat([semantic_code[:, :], predict_semantic], dim=-1) | |
prompt_semantic_code = semantic_code | |
return combine_semantic_code, prompt_semantic_code | |
def semantic2acoustic( | |
device, | |
combine_semantic_code, | |
acoustic_code, | |
n_timesteps=[25, 10, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1], | |
cfg=2.5, | |
rescale_cfg=0.75, | |
): | |
semantic_code = combine_semantic_code | |
cond = s2a_model_1layer.cond_emb(semantic_code) | |
prompt = acoustic_code[:, :, :] | |
predict_1layer = s2a_model_1layer.reverse_diffusion( | |
cond=cond, | |
prompt=prompt, | |
temp=1.5, | |
filter_thres=0.98, | |
n_timesteps=n_timesteps[:1], | |
cfg=cfg, | |
rescale_cfg=rescale_cfg, | |
) | |
cond = s2a_model_full.cond_emb(semantic_code) | |
prompt = acoustic_code[:, :, :] | |
predict_full = s2a_model_full.reverse_diffusion( | |
cond=cond, | |
prompt=prompt, | |
temp=1.5, | |
filter_thres=0.98, | |
n_timesteps=n_timesteps, | |
cfg=cfg, | |
rescale_cfg=rescale_cfg, | |
gt_code=predict_1layer, | |
) | |
vq_emb = codec_decoder.vq2emb(predict_full.permute(2, 0, 1), n_quantizers=12) | |
recovered_audio = codec_decoder(vq_emb) | |
prompt_vq_emb = codec_decoder.vq2emb(prompt.permute(2, 0, 1), n_quantizers=12) | |
recovered_prompt_audio = codec_decoder(prompt_vq_emb) | |
recovered_prompt_audio = recovered_prompt_audio[0][0].cpu().numpy() | |
recovered_audio = recovered_audio[0][0].cpu().numpy() | |
combine_audio = np.concatenate([recovered_prompt_audio, recovered_audio]) | |
return combine_audio, recovered_audio | |
# Load the model and checkpoints | |
def load_models(): | |
cfg_path = "./models/tts/maskgct/config/maskgct.json" | |
cfg = load_config(cfg_path) | |
semantic_model, semantic_mean, semantic_std = build_semantic_model(device) | |
semantic_codec = build_semantic_codec(cfg.model.semantic_codec, device) | |
codec_encoder, codec_decoder = build_acoustic_codec( | |
cfg.model.acoustic_codec, device | |
) | |
t2s_model = build_t2s_model(cfg.model.t2s_model, device) | |
s2a_model_1layer = build_s2a_model(cfg.model.s2a_model.s2a_1layer, device) | |
s2a_model_full = build_s2a_model(cfg.model.s2a_model.s2a_full, device) | |
# Download checkpoints | |
semantic_code_ckpt = hf_hub_download( | |
"amphion/MaskGCT", filename="semantic_codec/model.safetensors" | |
) | |
# codec_encoder_ckpt = hf_hub_download( | |
# "amphion/MaskGCT", filename="acoustic_codec/model.safetensors" | |
# ) | |
# codec_decoder_ckpt = hf_hub_download( | |
# "amphion/MaskGCT", filename="acoustic_codec/model_1.safetensors" | |
# ) | |
t2s_model_ckpt = hf_hub_download( | |
"amphion/MaskGCT", filename="t2s_model/model.safetensors" | |
) | |
s2a_1layer_ckpt = hf_hub_download( | |
"amphion/MaskGCT", filename="s2a_model/s2a_model_1layer/model.safetensors" | |
) | |
s2a_full_ckpt = hf_hub_download( | |
"amphion/MaskGCT", filename="s2a_model/s2a_model_full/model.safetensors" | |
) | |
safetensors.torch.load_model(semantic_codec, semantic_code_ckpt) | |
# safetensors.torch.load_model(codec_encoder, codec_encoder_ckpt) | |
# safetensors.torch.load_model(codec_decoder, codec_decoder_ckpt) | |
accelerate.load_checkpoint_and_dispatch(codec_encoder, "./acoustic_codec/model.safetensors") | |
accelerate.load_checkpoint_and_dispatch(codec_decoder, "./acoustic_codec/model_1.safetensors") | |
safetensors.torch.load_model(t2s_model, t2s_model_ckpt) | |
safetensors.torch.load_model(s2a_model_1layer, s2a_1layer_ckpt) | |
safetensors.torch.load_model(s2a_model_full, s2a_full_ckpt) | |
return ( | |
semantic_model, | |
semantic_mean, | |
semantic_std, | |
semantic_codec, | |
codec_encoder, | |
codec_decoder, | |
t2s_model, | |
s2a_model_1layer, | |
s2a_model_full, | |
) | |
def maskgct_inference( | |
prompt_speech_path, | |
prompt_text, | |
target_text, | |
language="en", | |
target_language="en", | |
target_len=None, | |
n_timesteps=25, | |
cfg=2.5, | |
rescale_cfg=0.75, | |
n_timesteps_s2a=[25, 10, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1], | |
cfg_s2a=2.5, | |
rescale_cfg_s2a=0.75, | |
device=torch.device("cuda:0"), | |
): | |
speech_16k = librosa.load(prompt_speech_path, sr=16000)[0] | |
speech = librosa.load(prompt_speech_path, sr=24000)[0] | |
# if prompt_text is None: | |
# prompt_text, language = get_prompt_text(prompt_speech_path) | |
combine_semantic_code, _ = text2semantic( | |
device, | |
speech_16k, | |
prompt_text, | |
language, | |
target_text, | |
target_language, | |
target_len, | |
n_timesteps, | |
cfg, | |
rescale_cfg, | |
) | |
acoustic_code = extract_acoustic_code(torch.tensor(speech).unsqueeze(0).to(device)) | |
_, recovered_audio = semantic2acoustic( | |
device, | |
combine_semantic_code, | |
acoustic_code, | |
n_timesteps=n_timesteps_s2a, | |
cfg=cfg_s2a, | |
rescale_cfg=rescale_cfg_s2a, | |
) | |
return recovered_audio | |
def inference( | |
prompt_wav, | |
prompt_text, | |
target_text, | |
target_len, | |
n_timesteps, | |
language, | |
target_language, | |
): | |
save_path = "./output/output.wav" | |
os.makedirs("./output", exist_ok=True) | |
recovered_audio = maskgct_inference( | |
prompt_wav, | |
prompt_text, | |
target_text, | |
language, | |
target_language, | |
target_len=target_len, | |
n_timesteps=int(n_timesteps), | |
device=device, | |
) | |
sf.write(save_path, recovered_audio, 24000) | |
return save_path | |
# Load models once | |
( | |
semantic_model, | |
semantic_mean, | |
semantic_std, | |
semantic_codec, | |
codec_encoder, | |
codec_decoder, | |
t2s_model, | |
s2a_model_1layer, | |
s2a_model_full, | |
) = load_models() | |
# Language list | |
language_list = ["en", "zh", "ja", "ko", "fr", "de"] | |
# Gradio interface | |
iface = gr.Interface( | |
fn=inference, | |
inputs=[ | |
gr.Audio(label="Upload Prompt Wav", type="filepath"), | |
gr.Textbox(label="Prompt Text"), | |
gr.Textbox(label="Target Text"), | |
gr.Number( | |
label="Target Duration (in seconds), if the target duration is less than 0, the system will estimate a duration.", value=-1 | |
), # Removed 'optional=True' | |
gr.Slider( | |
label="Number of Timesteps", minimum=15, maximum=100, value=25, step=1 | |
), | |
gr.Dropdown(label="Language", choices=language_list, value="en"), | |
gr.Dropdown(label="Target Language", choices=language_list, value="en"), | |
], | |
outputs=gr.Audio(label="Generated Audio"), | |
title="MaskGCT TTS Demo", | |
description=""" | |
## MaskGCT: Zero-Shot Text-to-Speech with Masked Generative Codec Transformer | |
[![arXiv](https://img.shields.io/badge/arXiv-Paper-COLOR.svg)](https://arxiv.org/abs/2409.00750) | |
[![hf](https://img.shields.io/badge/%F0%9F%A4%97%20HuggingFace-model-yellow)](https://huggingface.co/amphion/maskgct) | |
[![hf](https://img.shields.io/badge/%F0%9F%A4%97%20HuggingFace-demo-pink)](https://huggingface.co/spaces/amphion/maskgct) | |
[![readme](https://img.shields.io/badge/README-Key%20Features-blue)](https://github.com/open-mmlab/Amphion/tree/main/models/tts/maskgct) | |
""" | |
) | |
# Launch the interface | |
iface.launch(allowed_paths=["./output"]) | |