import torch import torchaudio import gradio as gr import soundfile as sf import wave import numpy as np from transformers import WhisperForCTC, WhisperProcessor, AutoModelForSeq2SeqLM, AutoTokenizer from transformers import OpenVoiceV2Processor, OpenVoiceV2 # Load ASR model and processor processor_asr = WhisperProcessor.from_pretrained("openai/whisper-large-v3") model_asr = WhisperForCTC.from_pretrained("openai/whisper-large-v3") # Load text-to-text model and tokenizer text_model = AutoModelForSeq2SeqLM.from_pretrained("meta-llama/Meta-Llama-3-8B") tokenizer = AutoTokenizer.from_pretrained("meta-llama/Meta-Llama-3-8B") # Load TTS model tts_processor = OpenVoiceV2Processor.from_pretrained("myshell-ai/OpenVoiceV2") tts_model = OpenVoiceV2.from_pretrained("myshell-ai/OpenVoiceV2") @spaces.GPU() # ASR function def transcribe(audio): waveform, sample_rate = torchaudio.load(audio) inputs = processor_asr(waveform, sampling_rate=sample_rate, return_tensors="pt", padding=True) with torch.no_grad(): logits = model_asr(inputs.input_values).logits predicted_ids = torch.argmax(logits, dim=-1) transcription = processor_asr.batch_decode(predicted_ids) return transcription[0] @spaces.GPU() # Text-to-text function def generate_response(text): inputs = tokenizer(text, return_tensors="pt", padding=True, truncation=True) outputs = text_model.generate(**inputs) response = tokenizer.decode(outputs[0], skip_special_tokens=True) return response @spaces.GPU() # TTS function def synthesize_speech(text): inputs = tts_processor(text, return_tensors="pt") with torch.no_grad(): mel_outputs, mel_outputs_postnet, _, alignments = tts_model.inference(inputs.input_ids) audio = tts_model.infer(mel_outputs_postnet) return audio @spaces.GPU() # Real-time processing function def real_time_pipeline(): # Adjust this part to handle live recording using soundfile and play back using simpleaudio import simpleaudio as sa import tempfile import time wake_word = "hello mate" wake_word_detected = False print("Listening for wake word...") with tempfile.NamedTemporaryFile(delete=False) as tmp_wav_file: tmp_wav_path = tmp_wav_file.name try: while True: # Capture audio here (this is a simplified example, you need actual audio capture logic) time.sleep(2) # Simulate 2 seconds of audio capture # Save the captured audio to the temp file for ASR data, sample_rate = sf.read(tmp_wav_path) sf.write(tmp_wav_path, data, sample_rate) # Step 1: Transcribe audio to text transcription = transcribe(tmp_wav_path).lower() if wake_word in transcription: wake_word_detected = True print("Wake word detected. Processing audio...") while wake_word_detected: # Capture audio here (this is a simplified example, you need actual audio capture logic) time.sleep(2) # Simulate 2 seconds of audio capture # Save the captured audio to the temp file for ASR data, sample_rate = sf.read(tmp_wav_path) sf.write(tmp_wav_path, data, sample_rate) # Step 1: Transcribe audio to text transcription = transcribe(tmp_wav_path) # Step 2: Generate response using text-to-text model response = generate_response(transcription) # Step 3: Synthesize speech from text synthesized_audio = synthesize_speech(response) # Save the synthesized audio to a temporary file output_path = "output.wav" torchaudio.save(output_path, synthesized_audio.squeeze(1), 22050) # Play the synthesized audio using simpleaudio wave_obj = sa.WaveObject.from_wave_file(output_path) play_obj = wave_obj.play() play_obj.wait_done() except KeyboardInterrupt: print("Stopping...") # Gradio interface gr_interface = gr.Interface( fn=real_time_pipeline, inputs=None, outputs=None, live=True, title="Real-Time Audio-to-Audio Model", description="ASR + Text-to-Text Model + TTS with Human-like Voice and Emotions" ) iface.launch(inline=False)