# # Copyright 2016 The BigDL Authors. # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. # import os import torch import time import intel_extension_for_pytorch as ipex import argparse import numpy as np import inquirer import sounddevice from bigdl.llm.transformers import AutoModelForCausalLM from bigdl.llm.transformers import AutoModelForSpeechSeq2Seq from transformers import LlamaTokenizer from transformers import WhisperProcessor from transformers import TextStreamer from colorama import Fore import speech_recognition as sr from datasets import load_dataset # you could tune the prompt based on your own model, # here the prompt tuning refers to https://huggingface.co/georgesung/llama2_7b_chat_uncensored#prompt-style DEFAULT_SYSTEM_PROMPT = """\ """ def get_prompt(message: str, chat_history: list[tuple[str, str]], system_prompt: str) -> str: texts = [f'[INST] <>\n{system_prompt}\n<>\n\n'] # The first user input is _not_ stripped do_strip = False for user_input, response in chat_history: user_input = user_input.strip() if do_strip else user_input do_strip = True texts.append(f'{user_input} [/INST] {response.strip()} [INST] ') message = message.strip() if do_strip else message texts.append(f'{message} [/INST]') return ''.join(texts) if __name__ == '__main__': parser = argparse.ArgumentParser(description='Predict Tokens using `generate()` API for Llama2 model') parser.add_argument('--llama2-repo-id-or-model-path', type=str, default="meta-llama/Llama-2-7b-chat-hf", help='The huggingface repo id for the Llama2 (e.g. `meta-llama/Llama-2-7b-chat-hf` and `meta-llama/Llama-2-13b-chat-hf`) to be downloaded' ', or the path to the huggingface checkpoint folder') parser.add_argument('--whisper-repo-id-or-model-path', type=str, default="openai/whisper-small", help='The huggingface repo id for the Whisper (e.g. `openai/whisper-small` and `openai/whisper-medium`) to be downloaded' ', or the path to the huggingface checkpoint folder') parser.add_argument('--n-predict', type=int, default=32, help='Max tokens to predict') args = parser.parse_args() # Select device mics = sr.Microphone.list_microphone_names() mics.insert(0, "Default") questions = [ inquirer.List('device_name', message="Which microphone do you choose?", choices=mics) ] answers = inquirer.prompt(questions) device_name = answers['device_name'] idx = mics.index(device_name) device_index = None if idx == 0 else idx - 1 print(f"The device name {device_name} is selected.") whisper_model_path = args.whisper_repo_id_or_model_path llama_model_path = args.llama2_repo_id_or_model_path dataset_path = "hf-internal-testing/librispeech_asr_dummy" # Load dummy dataset and read audio files ds = load_dataset(dataset_path, "clean", split="validation") print("Converting and loading models...") processor = WhisperProcessor.from_pretrained(whisper_model_path) # generate token ids whisper = AutoModelForSpeechSeq2Seq.from_pretrained(whisper_model_path, load_in_4bit=True, optimize_model=False) whisper.config.forced_decoder_ids = None whisper = whisper.to('xpu') llama_model = AutoModelForCausalLM.from_pretrained(llama_model_path, load_in_4bit=True, trust_remote_code=True, optimize_model=False) llama_model = llama_model.to('xpu') tokenizer = LlamaTokenizer.from_pretrained(llama_model_path) r = sr.Recognizer() with torch.inference_mode(): # warm up sample = ds[2]["audio"] input_features = processor(sample["array"], sampling_rate=sample["sampling_rate"], return_tensors="pt").input_features input_features = input_features.contiguous().to('xpu') torch.xpu.synchronize() predicted_ids = whisper.generate(input_features) torch.xpu.synchronize() output_str = processor.batch_decode(predicted_ids, skip_special_tokens=True) output_str = output_str[0] input_ids = tokenizer.encode(output_str, return_tensors="pt").to('xpu') output = llama_model.generate(input_ids, do_sample=False, max_new_tokens=32) output_str = tokenizer.decode(output[0], skip_special_tokens=True) torch.xpu.synchronize() with sr.Microphone(device_index=device_index, sample_rate=16000) as source: print("Calibrating...") r.adjust_for_ambient_noise(source, duration=5) while 1: print(Fore.YELLOW + "Listening now..." + Fore.RESET) try: audio = r.listen(source, timeout=5, phrase_time_limit=30) # refer to https://github.com/openai/whisper/blob/main/whisper/audio.py#L63 frame_data = np.frombuffer(audio.frame_data, np.int16).flatten().astype(np.float32) / 32768.0 print("Recognizing...") input_features = processor(frame_data, sampling_rate=audio.sample_rate, return_tensors="pt").input_features input_features = input_features.contiguous().to('xpu') except Exception as e: unrecognized_speech_text = ( f"Sorry, I didn't catch that. Exception was: \n {e}" ) print(unrecognized_speech_text) predicted_ids = whisper.generate(input_features) output_str = processor.batch_decode(predicted_ids, skip_special_tokens=True) output_str = output_str[0] print("\n" + Fore.GREEN + "Whisper : " + Fore.RESET + "\n" + output_str) print("\n" + Fore.BLUE + "BigDL-LLM: " + Fore.RESET) prompt = get_prompt(output_str, [], system_prompt=DEFAULT_SYSTEM_PROMPT) input_ids = tokenizer.encode(prompt, return_tensors="pt").to('xpu') streamer = TextStreamer(tokenizer, skip_special_tokens=True, skip_prompt=True) _ = llama_model.generate(input_ids, streamer=streamer, do_sample=False, max_new_tokens=args.n_predict)