import gradio as gr import torch import spaces import torchaudio from whisperspeech.vq_stoks import RQBottleneckTransformer from encodec.utils import convert_audio from transformers import AutoModelForCausalLM, AutoTokenizer, BitsAndBytesConfig, pipeline from transformers import StoppingCriteria, StoppingCriteriaList, TextIteratorStreamer from threading import Thread import logging import os from generate_audio import ( TTSProcessor, ) import uuid device = "cuda" if torch.cuda.is_available() else "cpu" vq_model = RQBottleneckTransformer.load_model( "whisper-vq-stoks-v3-7lang-fixed.model" ).to(device) # tts = TTSProcessor('cpu') use_8bit = False llm_path = "homebrewltd/Ichigo-llama3.1-s-instruct-v0.3-phase-3" tokenizer = AutoTokenizer.from_pretrained(llm_path) model_kwargs = {} if use_8bit: model_kwargs["quantization_config"] = BitsAndBytesConfig( load_in_8bit=True, llm_int8_enable_fp32_cpu_offload=False, llm_int8_has_fp16_weight=False, ) else: model_kwargs["torch_dtype"] = torch.bfloat16 model = AutoModelForCausalLM.from_pretrained(llm_path, **model_kwargs).to(device) @spaces.GPU def audio_to_sound_tokens_whisperspeech(audio_path): vq_model.ensure_whisper('cuda') wav, sr = torchaudio.load(audio_path) if sr != 16000: wav = torchaudio.functional.resample(wav, sr, 16000) with torch.no_grad(): codes = vq_model.encode_audio(wav.to(device)) codes = codes[0].cpu().tolist() result = ''.join(f'<|sound_{num:04d}|>' for num in codes) return f'<|sound_start|>{result}<|sound_end|>' @spaces.GPU def audio_to_sound_tokens_whisperspeech_transcribe(audio_path): vq_model.ensure_whisper('cuda') wav, sr = torchaudio.load(audio_path) if sr != 16000: wav = torchaudio.functional.resample(wav, sr, 16000) with torch.no_grad(): codes = vq_model.encode_audio(wav.to(device)) codes = codes[0].cpu().tolist() result = ''.join(f'<|sound_{num:04d}|>' for num in codes) return f'Transcribe the speech in this audio sample:<|sound_start|>{result}<|sound_end|>' # print(tokenizer.encode("<|sound_0001|>", add_special_tokens=False))# return the audio tensor # print(tokenizer.eos_token) @spaces.GPU def text_to_audio_file(text): # gen a random id for the audio file id = str(uuid.uuid4()) temp_file = f"./user_audio/{id}_temp_audio.wav" text = text text_split = "_".join(text.lower().split(" ")) # remove the last character if it is a period if text_split[-1] == ".": text_split = text_split[:-1] tts = TTSProcessor("cuda") tts.convert_text_to_audio_file(text, temp_file) # logging.info(f"Saving audio to {temp_file}") # torchaudio.save(temp_file, audio.cpu(), sample_rate=24000) print(f"Saved audio to {temp_file}") return temp_file @spaces.GPU def process_input(audio_file=None): for partial_message in process_audio(audio_file): yield partial_message @spaces.GPU def process_transcribe_input(audio_file=None): for partial_message in process_audio(audio_file, transcript=True): yield partial_message class StopOnTokens(StoppingCriteria): def __call__(self, input_ids: torch.LongTensor, scores: torch.FloatTensor, **kwargs) -> bool: # encode token stop_ids = [tokenizer.eos_token_id, 128009] # Adjust this based on your model's tokenizer for stop_id in stop_ids: if input_ids[0][-1] == stop_id: return True return False @spaces.GPU def process_audio(audio_file, transcript=False): if audio_file is None: raise ValueError("No audio file provided") logging.info(f"Audio file received: {audio_file}") logging.info(f"Audio file type: {type(audio_file)}") sound_tokens = audio_to_sound_tokens_whisperspeech_transcribe(audio_file) if transcript else audio_to_sound_tokens_whisperspeech(audio_file) logging.info("Sound tokens generated successfully") # logging.info(f"audio_file: {audio_file.name}") messages = [ {"role": "user", "content": sound_tokens}, ] stop = StopOnTokens() input_str = tokenizer.apply_chat_template(messages, tokenize=False) input_ids = tokenizer.encode(input_str, return_tensors="pt") input_ids = input_ids.to(model.device) streamer = TextIteratorStreamer(tokenizer, timeout=10., skip_prompt=True, skip_special_tokens=True) generation_kwargs = dict( input_ids=input_ids, streamer=streamer, max_new_tokens=1024, do_sample=False, stopping_criteria=StoppingCriteriaList([stop]) ) thread = Thread(target=model.generate, kwargs=generation_kwargs) thread.start() partial_message = "" for new_token in streamer: partial_message += new_token if tokenizer.eos_token in partial_message: break partial_message = partial_message.replace("assistant\n\n", "") yield partial_message # def stop_generation(): # # This is a placeholder. Implement actual stopping logic here if needed. # return "Generation stopped.", gr.Button.update(interactive=False) # take all the examples from the examples folder good_examples = [] for file in os.listdir("./examples"): if file.endswith(".wav"): good_examples.append([f"./examples/{file}"]) bad_examples = [] for file in os.listdir("./bad_examples"): if file.endswith(".wav"): bad_examples.append([f"./bad_examples/{file}"]) examples = [] examples.extend(good_examples) examples.extend(bad_examples) with gr.Blocks() as iface: gr.Markdown("# Ichigo-llama3-s: Llama3.1 with listening capabilities") gr.Markdown("Record your voice or upload audio and send it to the model.") gr.Markdown("Powered by [Homebrew Ltd](https://homebrew.ltd/) | [Read our blog post](https://homebrew.ltd/blog/llama3-just-got-ears)") with gr.Row(): input_type = gr.Radio(["text", "audio"], label="Input Type", value="audio") text_input = gr.Textbox(label="Send", visible=False) audio_input = gr.Audio(label="Audio", type="filepath", visible=True) # audio_output = gr.Audio(label="Converted Audio", type="filepath", visible=False) convert_button = gr.Button("Convert to Audio", visible=False) submit_button = gr.Button("Send") # transcrip_button = gr.Button("Make Model Transcribe the audio") text_output = gr.Textbox(label="Generated Text") def update_visibility(input_type): return (gr.update(visible=input_type == "text"), gr.update(visible=input_type == "text")) def convert_and_display(text): audio_file = text_to_audio_file(text) return audio_file def process_example(file_path): return update_visibility("audio") input_type.change( update_visibility, inputs=[input_type], outputs=[text_input, convert_button] ) convert_button.click( convert_and_display, inputs=[text_input], outputs=[audio_input] ) submit_button.click( process_input, inputs=[audio_input], outputs=[text_output] ) # transcrip_button.click( # process_transcribe_input, # inputs=[audio_input], # outputs=[text_output] # ) gr.Examples(examples, inputs=[audio_input]) iface.queue() iface.launch() # launch locally # iface.launch(server_name="0.0.0.0")