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import spaces | |
import torch | |
import gradio as gr | |
from transformers import pipeline, AutoModelForCausalLM, AutoTokenizer, AutoModel | |
import numpy as np | |
from espnet2.bin.tts_inference import Text2Speech | |
import yaml # Import yaml for config loading (though not used in the current code, kept for potential future use) | |
import os # Kept for potential future use (e.g., if loading config from files) | |
import requests # Corrected: Import the 'requests' library | |
import nltk # Import nltk | |
# Download required NLTK resources | |
try: | |
nltk.data.find('taggers/averaged_perceptron_tagger_eng') | |
except LookupError: | |
nltk.download('averaged_perceptron_tagger_eng') | |
try: | |
nltk.data.find('corpora/cmudict') # Check for cmudict | |
except LookupError: | |
nltk.download('cmudict') | |
# Load Whisper model | |
ASR_MODEL_NAME = "openai/whisper-medium.en" | |
asr_pipe = pipeline( | |
task="automatic-speech-recognition", | |
model=ASR_MODEL_NAME, | |
chunk_length_s=30, | |
device='cuda' if torch.cuda.is_available() else 'cpu', # Use GPU if available | |
) | |
all_special_ids = asr_pipe.tokenizer.all_special_ids | |
transcribe_token_id = all_special_ids[-5] | |
translate_token_id = all_special_ids[-6] | |
def _preload_and_load_models(): | |
global vicuna_tokenizer, vicuna_model | |
VICUNA_MODEL_NAME = "EleutherAI/gpt-neo-2.7B" # Or another model | |
vicuna_tokenizer = AutoTokenizer.from_pretrained(VICUNA_MODEL_NAME) | |
vicuna_model = AutoModelForCausalLM.from_pretrained( | |
VICUNA_MODEL_NAME, | |
torch_dtype=torch.float16, | |
device_map="auto", # or.to('cuda') | |
) #.to('cuda') # Explicitly move to CUDA after loading | |
_preload_and_load_models() | |
tts = Text2Speech.from_pretrained("espnet/kan-bayashi_ljspeech_vits") | |
def process_audio(microphone, state, task="transcribe"): | |
if microphone is None: | |
return state, state, None | |
asr_pipe.model.config.forced_decoder_ids = [ | |
[2, transcribe_token_id if task == "transcribe" else translate_token_id] | |
] | |
text = asr_pipe(microphone)["text"] | |
system_prompt = """You are a friendly and enthusiastic tutor for young children (ages 6-9). | |
You answer questions clearly and simply, using age-appropriate language. | |
You are also a little bit silly and like to make jokes.""" | |
prompt = f"{system_prompt}\nUser: {text}" | |
with torch.no_grad(): | |
vicuna_input = vicuna_tokenizer(prompt, return_tensors="pt").to('cuda') | |
vicuna_output = vicuna_model.generate(**vicuna_input, max_new_tokens=192) | |
vicuna_response = vicuna_tokenizer.decode(vicuna_output[0], skip_special_tokens=True) # Access the first sequence [0] | |
vicuna_response = vicuna_response.replace(prompt, "").strip() | |
updated_state = state + "\nUser: " + text + "\n" + "Tutor: " + vicuna_response # Include user input in state | |
try: | |
with torch.no_grad(): | |
# The espnet TTS model outputs a dictionary | |
output = tts(vicuna_response) | |
wav = output["wav"] | |
sr = tts.fs # Get the sampling rate from the tts object | |
audio_arr = wav.cpu().numpy() | |
SAMPLE_RATE = sr | |
audio_arr = audio_arr / np.abs(audio_arr).max() # Normalize to -1 to 1 | |
audio_output = (SAMPLE_RATE, audio_arr) | |
#sf.write('generated_audio.wav', audio_arr, SAMPLE_RATE) # Removed writing to file | |
except requests.exceptions.RequestException as e: | |
print(f"Error in Hugging Face API request: {e}") | |
audio_output = None | |
except Exception as e: | |
print(f"Error in speech synthesis: {e}") | |
audio_output = None | |
return updated_state, updated_state, audio_output | |
with gr.Blocks(title="Whisper, Vicuna, & TTS Demo") as demo: # Updated title | |
gr.Markdown("# Speech-to-Text-to-Speech Demo with Vicuna and Hugging Face TTS") # Updated Markdown | |
gr.Markdown("Speak into your microphone, get a transcription, Vicuna will process it, and then you'll hear the result!") | |
with gr.Tab("Transcribe & Synthesize"): | |
mic_input = gr.Audio(sources="microphone", type="filepath", label="Speak Here") | |
transcription_output = gr.Textbox(lines=5, label="Transcription and Vicuna Response") | |
audio_output = gr.Audio(label="Synthesized Speech", type="numpy") # Important: type="numpy" | |
transcription_state = gr.State(value="") | |
mic_input.change( | |
fn=process_audio, # Call the combined function | |
inputs=[mic_input, transcription_state, gr.Radio(["transcribe", "translate"], label="Task", value="transcribe")], | |
outputs=[transcription_output, transcription_state, audio_output] | |
) | |
demo.launch(share=False) |