from transformers import pipeline import torch device = "cuda:0" if torch.cuda.is_available() else "cpu" classifier = pipeline( "audio-classification", model="MIT/ast-finetuned-speech-commands-v2", device=device ) from transformers.pipelines.audio_utils import ffmpeg_microphone_live def launch_fn( wake_word="marvin", prob_threshold=0.5, chunk_length_s=2.0, stream_chunk_s=0.25, debug=False, ): if wake_word not in classifier.model.config.label2id.keys(): raise ValueError( f"Wake word {wake_word} not in set of valid class labels, pick a wake word in the set {classifier.model.config.label2id.keys()}." ) sampling_rate = classifier.feature_extractor.sampling_rate mic = ffmpeg_microphone_live( sampling_rate=sampling_rate, chunk_length_s=chunk_length_s, stream_chunk_s=stream_chunk_s, ) print("Listening for wake word...") for prediction in classifier(mic): prediction = prediction[0] if debug: print(prediction) if prediction["label"] == wake_word: if prediction["score"] > prob_threshold: return True transcriber = pipeline( "automatic-speech-recognition", model="openai/whisper-base.en", device=device ) import sys def transcribe(chunk_length_s=5.0, stream_chunk_s=1.0): sampling_rate = transcriber.feature_extractor.sampling_rate mic = ffmpeg_microphone_live( sampling_rate=sampling_rate, chunk_length_s=chunk_length_s, stream_chunk_s=stream_chunk_s, ) print("Start speaking...") for item in transcriber(mic, generate_kwargs={"max_new_tokens": 128}): sys.stdout.write("\033[K") print(item["text"], end="\r") if not item["partial"][0]: break return item["text"] from huggingface_hub import HfFolder import requests def query(text, model_id="tiiuae/falcon-7b-instruct"): api_url = f"https://api-inference.huggingface.co/models/{model_id}" headers = {"Authorization": f"Bearer {HfFolder().get_token()}"} payload = {"inputs": text} print(f"Querying...: {text}") response = requests.post(api_url, headers=headers, json=payload) return response.json()[0]["generated_text"][len(text) + 1 :] from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts") model = SpeechT5ForTextToSpeech.from_pretrained("microsoft/speecht5_tts").to(device) vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device) from datasets import load_dataset embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0) def synthesise(text): inputs = processor(text=text, return_tensors="pt") speech = model.generate_speech( inputs["input_ids"].to(device), speaker_embeddings.to(device), vocoder=vocoder ) return speech.cpu() if __name__ == "__main__": launch_fn(debug=True) # transcription = transcribe() # response = query(transcription) # audio = synthesise(response) # Audio(audio, rate=16000, autoplay=True)