removed transforms AutoModelForSpeechSeq2Seq, AutoProcessor

app.py

@@ -11,7 +11,7 @@ def audio_converter(audio_file:str):
     return audio_wav_filename
 
 def asr_transcriber(audio_file):
-    from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
+    from transformers import pipeline
     import torch
     import optimum
 
@@ -29,7 +29,7 @@ def asr_transcriber(audio_file):
     #device_id = "mps"
     #torch_dtype = torch.float16
     flash = False
-    ts = True
+    ts = False
 
     #Try to optimize when CPU and float32
     model_id = "openai/whisper-tiny"