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import gradio as gr | |
from transformers import pipeline | |
import numpy as np | |
transcriber = pipeline("automatic-speech-recognition", model="openai/whisper-base.en") | |
def transcribe(stream, new_chunk): | |
sr, y = new_chunk | |
y = y.astype(np.float32) | |
y /= np.max(np.abs(y)) | |
if stream is not None: | |
stream = np.concatenate([stream, y]) | |
else: | |
stream = y | |
return stream, transcriber({"sampling_rate": sr, "raw": stream})["text"] , new_chunk | |
demo = gr.Interface( | |
transcribe, | |
["state", gr.Audio(sources=["microphone"], streaming=True)], | |
["state", "text", gr.Audio(label="Output", streaming=True, autoplay=True)], | |
live=True, | |
) | |
demo.launch() | |
# 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) |