import spaces import gradio as gr import torch import soundfile as sf from transformers import AutoTokenizer, AutoModelForCausalLM from xcodec2.modeling_xcodec2 import XCodec2Model import tempfile import torchaudio import os device = "cuda" if torch.cuda.is_available() else "cpu" #################### # Global model loading #################### model_name = "fakeavatar/vtubers-4" print("Loading tokenizer & model ...") tokenizer = AutoTokenizer.from_pretrained(model_name) model = AutoModelForCausalLM.from_pretrained(model_name) if os.name != "nt" and torch.cuda.is_available(): # 'nt' means Windows, so this runs on Linux/macOS model = torch.compile(model) torch.backends.cudnn.benchmark = True # For variable input sizes torch.backends.cuda.matmul.allow_tf32 = True # Allow TF32 on Ampere GPUs model.eval().to(device) print("Loading XCodec2Model ...") codec_model_path = "HKUSTAudio/xcodec2" Codec_model = XCodec2Model.from_pretrained(codec_model_path) Codec_model.eval().to(device) print("Models loaded.") #################### # Inference function #################### def extract_speech_ids(speech_tokens_str): """ Restore an integer 23456 from tokens like <|s_23456|> """ speech_ids = [] for token_str in speech_tokens_str: if token_str.startswith("<|s_") and token_str.endswith("|>"): num_str = token_str[4:-2] num = int(num_str) speech_ids.append(num) else: print(f"Unexpected token: {token_str}") return speech_ids @spaces.GPU def text2speech(input_text, num_samples): """ Convert text to speech waveform and return the audio file path """ results = [] with torch.no_grad(): audio, sr = torchaudio.load("./sample.wav") vq_code = Codec_model.encode_code(audio.to("cuda")) vq_strings = [f"<|s_{i}|>" for i in vq_code.to("cpu")[0][0].tolist()] vq_str = "".join(vq_strings) for i in range(0, num_samples): # Add start and end tokens around the input text formatted_text = f"<|TEXT_UNDERSTANDING_START|>{input_text}<|TEXT_UNDERSTANDING_END|>" chat = [ {"role": "user", "content": "Convert the text to speech:" + formatted_text}, {"role": "assistant", "content": f"<|SPEECH_GENERATION_START|>"} ] chat = [ {"role": "system", "content": "the speaker is yui. She has a mild chinese accent and is speaking english. The voice is flowing and nasal, high pitched with a measured speed. The sound is recorded in a fairly clean and carries a medium happy emotion."}, {"role": "user", "content": "Convert the text to speech:" + f"<|TEXT_UNDERSTANDING_START|>Hey, wake up! {input_text}<|TEXT_UNDERSTANDING_END|>"}, {"role": "assistant", "content": "<|SPEECH_GENERATION_START|>" + vq_str}, # {"role": "user", "content": formatted_text}, # {"role": "assistant", "content": "<|SPEECH_GENERATION_START|>"} ] # tokenizer.apply_chat_template is used in the Llasa-style dialogue model input_ids = tokenizer.apply_chat_template( chat, tokenize=True, return_tensors='pt', continue_final_message=True ).to(device) # End token speech_end_id = tokenizer.convert_tokens_to_ids("<|SPEECH_GENERATION_END|>") # Text generation outputs = model.generate( input_ids, max_length=2048, # We trained our model with a max length of 2048 eos_token_id=speech_end_id, do_sample=True, top_p=0.95, # Adjusts the diversity of generated content temperature=0.9, # Controls randomness in output repetition_penalty=1.2, ) # Extract newly generated tokens (excluding the input part) generated_ids = outputs[0][input_ids.shape[1]:-1] if (generated_ids.shape[0] < 2): continue speech_tokens_str = tokenizer.batch_decode(generated_ids, skip_special_tokens=True) # Extract <|s_23456|> as [23456 ...] speech_tokens_int = extract_speech_ids(speech_tokens_str) speech_tokens_int = torch.tensor(speech_tokens_int).to(device).unsqueeze(0).unsqueeze(0) # Decode waveform using XCodec2Model gen_wav = Codec_model.decode_code(speech_tokens_int) # [batch, channels, samples] # Get audio data and sample rate audio = gen_wav[0, 0, :].cpu().numpy() sample_rate = 16000 # Save the audio to a temporary file with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmpfile: sf.write(tmpfile.name, audio, sample_rate) audio_path = tmpfile.name results.append(audio_path) while len(results) < 10: results.append(results[-1]) return results #################### # Gradio Interface #################### # Slider to control the number of audio samples to generate num_samples_slider = gr.Slider(minimum=1, maximum=10, value=4, step=1, label="Number of Audio Samples") demo = gr.Interface( fn=text2speech, inputs=[gr.Textbox(label="Enter text", lines=5), num_samples_slider], outputs=[gr.Audio(label=f"Generated Audio {i+1}", type="numpy") for i in range(10)], title="VTuber TTS", description="Input a piece of text in English, and click to generate speech." ) if __name__ == "__main__": demo.launch( share=True )