Update app.py

app.py

@@ -8,7 +8,7 @@ from pydub import AudioSegment
 from IPython.display import Audio
 import os
 import accelerate
-import pyaudio
+#import pyaudio
 import numpy as np
 
 # Create PyAudio object
@@ -18,7 +18,7 @@ SAMPLING_RATE = 16000
 vocals_data = bytes()
 
 # preprocess and crop audio file
-def audio_preprocess(input_file, in_data, frame_count, time_info, status):
+def audio_preprocess(input_file): #, in_data, frame_count, time_info, status):
 
 # Define callback function for audio processing
 
@@ -32,10 +32,14 @@ def audio_preprocess(input_file, in_data, frame_count, time_info, status):
     
     # Convert vocals to audio data
     vocals_data = vocals['vocals'].flatten().astype(np.int16).tobytes()
+    separated_audio = vocals_data
     
     # Return processed data for output
-    return vocals_data, pyaudio.paContinue
+    return vocals_data, pyaudio.paContinue, processed_audio
 
+    
+# audio processing 2?
+def py_audio():
     # Open stream for recording
     stream = p.open(format=pyaudio.paInt16, channels=1, rate=SAMPLING_RATE, input=True, output=True,
                     frames_per_buffer=CHUNK_SIZE, stream_callback=process_audio)
@@ -80,8 +84,6 @@ def audio_preprocess(input_file, in_data, frame_count, time_info, status):
     return processed_audio
 
 
-
-
 # ASR transcription
 def asr_model(processed_audio):
    # load audio file
@@ -132,7 +134,7 @@ def senti_model(transcription):
 def main(input_file):
 
 
-   processed_audio = audio_preprocess(input_file)
+   separated_audio = audio_preprocess(input_file)
    transcription = asr_model(processed_audio)
    final_result = senti_model(transcription)
    st.write(final_result)