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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 whisper
import langid
processor = SeamlessM4TFeatureExtractor.from_pretrained("facebook/w2v-bert-2.0")
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
whisper_model = whisper.load_model("turbo")
def detect_speech_language(speech_file):
# load audio and pad/trim it to fit 30 seconds
audio = whisper.load_audio(speech_file)
audio = whisper.pad_or_trim(audio)
# make log-Mel spectrogram and move to the same device as the model
mel = whisper.log_mel_spectrogram(audio, n_mels=128).to(whisper_model.device)
# detect the spoken language
_, probs = whisper_model.detect_language(mel)
return max(probs, key=probs.get)
def detect_text_language(text):
return langid.classify(text)[0]
@torch.no_grad()
def get_prompt_text(speech_16k, language):
full_prompt_text = ""
shot_prompt_text = ""
short_prompt_end_ts = 0.0
asr_result = whisper_model.transcribe(speech_16k, language=language)
full_prompt_text = asr_result["text"] # whisper asr result
#text = asr_result["segments"][0]["text"] # whisperx asr result
shot_prompt_text = ""
short_prompt_end_ts = 0.0
for segment in asr_result["segments"]:
shot_prompt_text = shot_prompt_text + segment['text']
short_prompt_end_ts = segment['end']
if short_prompt_end_ts >= 4:
break
return full_prompt_text, shot_prompt_text, short_prompt_end_ts
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
@torch.no_grad()
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
@torch.no_grad()
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
@torch.no_grad()
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
@torch.no_grad()
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
@torch.no_grad()
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,
)
@torch.no_grad()
def maskgct_inference(
prompt_speech_path,
target_text,
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]
prompt_language = detect_speech_language(prompt_speech_path)
full_prompt_text, short_prompt_text, shot_prompt_end_ts = get_prompt_text(prompt_speech_path,
prompt_language)
# use the first 4+ seconds wav as the prompt in case the prompt wav is too long
speech = speech[0: int(shot_prompt_end_ts * 24000)]
speech_16k = speech_16k[0: int(shot_prompt_end_ts*16000)]
target_language = detect_text_language(target_text)
combine_semantic_code, _ = text2semantic(
device,
speech_16k,
short_prompt_text,
prompt_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
@spaces.GPU
def inference(
prompt_wav,
target_text,
target_len,
n_timesteps,
):
save_path = "./output/output.wav"
os.makedirs("./output", exist_ok=True)
recovered_audio = maskgct_inference(
prompt_wav,
target_text,
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="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
),
],
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"])
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