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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() | |
transcription = transcribe() | |
response = query(transcription) | |
audio = synthesise(response) | |
Audio(audio, rate=16000, autoplay=True) |