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Update app.py
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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)