import copy import math import os import sys import gradio as gr import numpy as np import torch from numba import jit from transformers import AutoConfig, AutoModelForCausalLM, AutoTokenizer, TextIteratorStreamer from transformers.generation import GenerationConfig from vita_audio.data.processor.audio_processor import add_audio_input_contiguous from vita_audio.tokenizer import get_audio_tokenizer PUNCTUATION = "!?。"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏." @jit def float_to_int16(audio: np.ndarray) -> np.ndarray: am = int(math.ceil(float(np.abs(audio).max())) * 32768) am = 32767 * 32768 // am return np.multiply(audio, am).astype(np.int16) def is_wav(file_path): wav_extensions = {".wav"} _, ext = os.path.splitext(file_path) return ext.lower() in wav_extensions def _parse_text(text): lines = text.split("\n") lines = [line for line in lines if line != ""] count = 0 for i, line in enumerate(lines): if "```" in line: count += 1 items = line.split("`") if count % 2 == 1: lines[i] = f'
'
            else:
                lines[i] = "
" else: if i > 0 and count % 2 == 1: line = line.replace("`", r"\`") line = line.replace("<", "<") line = line.replace(">", ">") line = line.replace(" ", " ") line = line.replace("*", "*") line = line.replace("_", "_") line = line.replace("-", "-") line = line.replace(".", ".") line = line.replace("!", "!") line = line.replace("(", "(") line = line.replace(")", ")") line = line.replace("$", "$") lines[i] = "
" + line return "".join(lines) def _launch_demo(model, tokenizer, audio_tokenizer): def predict(_chatbot, task_history, task): chat_query = task_history[-1][0] print(task_history) messages = [] audio_path_list = [] if task == "Spoken QA": messages = [ { "role": "system", # "content": "Your Name: Luke\nYour Gender: male\n\nRespond in a text-audio interleaved manner.", # "content": "Your Name: Lucy\nYour Gender: female\nRespond in a text-audio interleaved manner.", "content": "Your Name: Omni\nYour Gender: female\nRespond in a text-audio interleaved manner.", }, ] for i, (q, a) in enumerate(task_history): if isinstance(q, (tuple, list)) and is_wav(q[0]): audio_path_list.append(q[0]) messages = messages + [ { "role": "user", "content": f"\n<|audio|>", }, ] else: messages = messages + [ { "role": "user", "content": q, }, ] if a != None: messages = messages + [ { "role": "assistant", "content": a, }, ] model.generation_config.do_sample = False elif task == "TTS": for i, (q, a) in enumerate(task_history): if isinstance(q, (tuple, list)) and is_wav(q[0]): audio_path_list.append(q[0]) messages = messages + [ { "role": "user", "content": f"\n<|audio|>", }, ] else: messages = messages + [ { "role": "user", "content": f"Convert the text to speech.\n{q}", }, ] if a != None: messages = messages + [ { "role": "assistant", "content": a, }, ] model.generation_config.do_sample = True elif task == "ASR": for i, (q, a) in enumerate(task_history): if isinstance(q, (tuple, list)) and is_wav(q[0]): audio_path_list.append(q[0]) messages = messages + [ { "role": "user", "content": f"Convert the speech to text.\n<|audio|>", }, ] else: messages = messages + [ { "role": "user", "content": f"{q}", }, ] if a != None: messages = messages + [ { "role": "assistant", "content": a, }, ] model.generation_config.do_sample = False add_generation_prompt = True input_ids = tokenizer.apply_chat_template( messages, tokenize=True, add_generation_prompt=add_generation_prompt, # return_tensors="pt", ) input_ids, audios, audio_indices = add_audio_input_contiguous( input_ids, audio_path_list, tokenizer, audio_tokenizer ) input_ids = torch.tensor([input_ids], dtype=torch.long).to("cuda") # print("input", tokenizer.decode(input_ids[0], skip_special_tokens=False), flush=True) if audio_path_list == []: audios = None audio_indices = None outputs = model.generate( input_ids, audios=audios, audio_indices=audio_indices, ) output = tokenizer.decode(outputs[0], skip_special_tokens=False) # print(f"{output=}", flush=True) audio_offset = tokenizer.convert_tokens_to_ids("<|audio_0|>") begin_of_audio = tokenizer.convert_tokens_to_ids("<|begin_of_audio|>") end_of_audio = tokenizer.convert_tokens_to_ids("<|end_of_audio|>") im_end = tokenizer.convert_tokens_to_ids("<|im_end|>") response = outputs[0][len(input_ids[0]) :] audio_tokens = [] text_tokens = [] for token_id in response: if token_id >= audio_offset: audio_tokens.append(token_id - audio_offset) elif ( (token_id.item() != begin_of_audio) and (token_id.item() != end_of_audio) and (token_id.item() != im_end) ): text_tokens.append(token_id) if len(audio_tokens) > 0: tts_speech = audio_tokenizer.decode(audio_tokens) audio_np = float_to_int16(tts_speech.cpu().numpy()) tts_speech = (22050, audio_np) else: tts_speech = None # import pdb;pdb.set_trace() history_response = tokenizer.decode(text_tokens) task_history[-1] = (chat_query, history_response) _chatbot[-1] = (chat_query, history_response) # print("query",chat_query) # print("task_history",task_history) # print(_chatbot) # print("answer: ",outputs) return _chatbot, tts_speech def add_text(history, task_history, text): task_text = text # import pdb;pdb.set_trace() if len(text) >= 2 and text[-1] in PUNCTUATION and text[-2] not in PUNCTUATION: task_text = text[:-1] history = history + [(_parse_text(text), None)] task_history = task_history + [(task_text, None)] return history, task_history, "" def add_audio(history, task_history, file): print(file) if file is None: return history, task_history history = history + [((file,), None)] task_history = task_history + [((file,), None)] return history, task_history def reset_user_input(): # import pdb;pdb.set_trace() return gr.update(value="") def reset_state(task_history): task_history.clear() return [] with gr.Blocks(title="VITA-Audio-Plus-Vanilla") as demo: gr.Markdown("""
VITA-Audio-Plus-Vanilla
""") gr.Markdown( """
The deployment of the VITA-Audio-Plus-Vanilla model employs a non-streaming deployment approach. The currently deployed model is VITA-Audio-Plus-Vanilla. For the ASR and TTS tasks, only single-turn dialogues are supported. In the Spoken QA task, generated text is used as dialogue history to reduce the context length.
""" ) chatbot = gr.Chatbot( label="VITA-Audio-Plus-Vanilla", elem_classes="control-height", height=500 ) query = gr.Textbox(lines=2, label="Text Input") task_history = gr.State([]) with gr.Row(): add_text_button = gr.Button("Submit Text (提交文本)") add_audio_button = gr.Button("Submit Audio (提交音频)") empty_bin = gr.Button("🧹 Clear History (清除历史)") task = gr.Radio(choices=["ASR", "TTS", "Spoken QA"], label="TASK", value="Spoken QA") with gr.Row(scale=1): record_btn = gr.Audio( sources=["microphone", "upload"], type="filepath", label="🎤 Record or Upload Audio (录音或上传音频)", show_download_button=True, waveform_options=gr.WaveformOptions(sample_rate=16000), ) audio_output = gr.Audio( label="Play", streaming=True, autoplay=True, show_download_button=True ) add_text_button.click( add_text, [chatbot, task_history, query], [chatbot, task_history], show_progress=True ).then(reset_user_input, [], [query]).then( predict, [chatbot, task_history, task], [chatbot, audio_output], show_progress=True ) empty_bin.click(reset_state, [task_history], [chatbot], show_progress=True) add_audio_button.click( add_audio, [chatbot, task_history, record_btn], [chatbot, task_history], show_progress=True, ).then(predict, [chatbot, task_history, task], [chatbot, audio_output], show_progress=True) server_port = 18806 demo.launch( share=False, debug=True, server_name="0.0.0.0", server_port=server_port, show_api=False, show_error=False, ) def main(): model_name_or_path = "VITA-MLLM/VITA-Audio-Plus-Vanilla" device_map = "cuda:0" sys.path.append("third_party/GLM-4-Voice/") sys.path.append("third_party/GLM-4-Voice/cosyvoice/") sys.path.append("third_party/GLM-4-Voice/third_party/Matcha-TTS/") from huggingface_hub import snapshot_download audio_tokenizer_path = snapshot_download(repo_id="THUDM/glm-4-voice-tokenizer") flow_path = snapshot_download(repo_id="THUDM/glm-4-voice-decoder") audio_tokenizer_rank = 0 audio_tokenizer_type = "sensevoice_glm4voice" torch_dtype = torch.bfloat16 audio_tokenizer = get_audio_tokenizer( audio_tokenizer_path, audio_tokenizer_type, flow_path=flow_path, rank=audio_tokenizer_rank ) from evaluation.get_chat_template import qwen2_chat_template as chat_template tokenizer = AutoTokenizer.from_pretrained( model_name_or_path, trust_remote_code=True, chat_template=chat_template, ) # print(f"{tokenizer=}") # print(f"{tokenizer.get_chat_template()=}") model = AutoModelForCausalLM.from_pretrained( model_name_or_path, trust_remote_code=True, device_map=device_map, torch_dtype=torch_dtype, attn_implementation="flash_attention_2", ).eval() # print(f"{model.config.model_type=}") model.generation_config = GenerationConfig.from_pretrained( model_name_or_path, trust_remote_code=True ) model.generation_config.max_new_tokens = 4096 model.generation_config.chat_format = "chatml" model.generation_config.max_window_size = 8192 model.generation_config.use_cache = True model.generation_config.do_sample = True model.generation_config.temperature = 1.0 model.generation_config.top_k = 50 model.generation_config.top_p = 1.0 model.generation_config.num_beams = 1 model.generation_config.pad_token_id = tokenizer.pad_token_id model.generation_config.mtp_inference_mode = [8192, 10] _launch_demo(model, tokenizer, audio_tokenizer) if __name__ == "__main__": main()