update code

app.py

@@ -56,40 +56,63 @@ class AudioProcessor:
     
     def _process_with_ffmpeg(self, audio_input):
         # Your existing FFmpeg processing
-        pass
-        
-    def _process_with_sounddevice(self, audio_input):
-        # Process using sounddevice
-        duration = 5  # seconds
-        print(f"Recording with sounddevice (rate={self.sample_rate})...")
-        audio_data = sd.rec(int(duration * self.sample_rate), 
-                          samplerate=self.sample_rate, 
-                          channels=1)
-        sd.wait()
-        return (audio_data.flatten(), self.sample_rate)
-        
-    def _process_with_librosa(self, audio_input):
-        # Process using librosa
-        import librosa
         if isinstance(audio_input, tuple):
             return audio_input
-        elif isinstance(audio_input, str):
-            return librosa.load(audio_input, sr=self.sample_rate)
-        else:
-            # Handle other input types
+        try:
+            import ffmpeg
+            # Process audio file with ffmpeg
             with tempfile.NamedTemporaryFile(suffix='.wav', delete=False) as tmp:
                 tmp.write(audio_input)
                 tmp.flush()
-                data, sr = librosa.load(tmp.name, sr=self.sample_rate)
+                out, _ = (
+                    ffmpeg.input(tmp.name)
+                    .output('pipe:', format='f32le', ac=1, ar=self.sample_rate)
+                    .run(capture_stdout=True)
+                )
                 os.unlink(tmp.name)
-                return (data, sr)
+                return (np.frombuffer(out, dtype=np.float32), self.sample_rate)
+        except Exception as e:
+            raise Exception(f"FFmpeg processing failed: {str(e)}")
+        
+    def _process_with_sounddevice(self, audio_input):
+        # Process using sounddevice
+        if isinstance(audio_input, tuple):
+            return audio_input
+        try:
+            duration = 5  # seconds
+            print(f"Recording with sounddevice (rate={self.sample_rate})...")
+            audio_data = sd.rec(int(duration * self.sample_rate), 
+                              samplerate=self.sample_rate, 
+                              channels=1)
+            sd.wait()
+            return (audio_data.flatten(), self.sample_rate)
+        except Exception as e:
+            raise Exception(f"Sounddevice processing failed: {str(e)}")
+        
+    def _process_with_librosa(self, audio_input):
+        # Process using librosa
+        try:
+            import librosa
+            if isinstance(audio_input, tuple):
+                return audio_input
+            elif isinstance(audio_input, str):
+                return librosa.load(audio_input, sr=self.sample_rate)
+            else:
+                # Handle other input types
+                with tempfile.NamedTemporaryFile(suffix='.wav', delete=False) as tmp:
+                    tmp.write(audio_input)
+                    tmp.flush()
+                    data, sr = librosa.load(tmp.name, sr=self.sample_rate)
+                    os.unlink(tmp.name)
+                    return (data, sr)
+        except Exception as e:
+            raise Exception(f"Librosa processing failed: {str(e)}")
     
     def _process_fallback(self, audio_input):
-        # Simple numpy fallback
+        # Simple numpy fallback with proper error handling
         if isinstance(audio_input, tuple):
             return audio_input
-        return (np.random.random(16000), 16000  # Mock data
-
+        return (np.random.random(16000), 16000)  # Now properly closed
 # Modified Interface with Audio Debugging
 def create_debug_interface():
     audio_processor = AudioProcessor()