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import time | |
from fastapi import FastAPI, File | |
from faster_whisper import WhisperModel | |
from utils import ffmpeg_read, stt | |
from sentence_transformers import SentenceTransformer, util | |
import torch | |
app = FastAPI() | |
whisper_models = ["tiny", "base", "small", "medium", "large-v1", "large-v2"] | |
audio_model = WhisperModel("base", compute_type="int8", device="cpu") | |
text_model = SentenceTransformer('all-MiniLM-L6-v2') | |
corpus_embeddings = torch.load('corpus_embeddings.pt') | |
def speech_to_text(upload_audio, model_type="whisper"): | |
""" | |
Transcribe audio using whisper model. | |
""" | |
audio_path = ffmpeg_read(upload_audio, sampling_rate=16000) | |
# Transcribe audio | |
if model_type == "whisper": | |
transcribe_options = dict(task="transcribe", language="ja", beam_size=5, best_of=5, vad_filter=True) | |
segments_raw, info = audio_model.transcribe(audio_path, **transcribe_options) | |
segments = [segment.text for segment in segments_raw] | |
return ' '.join(segments) | |
else: | |
text = stt(audio_path) | |
return text | |
def read_root(): | |
return {"Message": "Application startup complete"} | |
async def voice_detect_api( | |
voice_input: bytes = File(None), | |
threshold: float = 0.8, | |
model_type: str = "whisper" | |
): | |
""" | |
API to detect voice from audio file. | |
""" | |
start = time.time() | |
text = speech_to_text(voice_input, model_type) | |
query_embedding = text_model.encode(text, convert_to_tensor=True) | |
hits = util.semantic_search(query_embedding, corpus_embeddings, top_k=1)[0] | |
if hits[0]['score'] > threshold: | |
similar = 1 | |
else: | |
similar = 0 | |
end = time.time() | |
return {"text": text, | |
"similar": similar, | |
"time_taken": end - start} | |