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)