diff --git a/vz_acoustic_scene_analysis/scripts/ros_interface.py b/vz_acoustic_scene_analysis/scripts/ros_interface.py
index 2617362..5d68206 100755
--- a/vz_acoustic_scene_analysis/scripts/ros_interface.py
+++ b/vz_acoustic_scene_analysis/scripts/ros_interface.py
@@ -3,14 +3,13 @@
 # Non ROS imports
 from __future__ import print_function
 import pyaudio
-import wave
 import numpy as np
-from sympy import re
 import usb.core
 import struct
-import time
+import scipy
 import sys
 import os
+import os.path
 import pickle
 from contextlib import contextmanager
 
@@ -213,21 +212,27 @@ class ROSInterface:
         self.loaded_model = pickle.load(open(self.cough_classifier_file, 'rb'))
         self.loaded_scaler = pickle.load(open(self.cough_classifier_scaler, 'rb'))
 
+        # for testing audio collection
+        # self.file_name = 'testing_audio_node1.wav'
+        # self.dir = os.path.join('/home/hello-robot/clone/Robot_Autonomous_Navigation/catkin_ws/src/stretch_ros/vz_acoustic_scene_analysis/scripts/', self.file_name)
+
 
 
     def get_audio(self):
-        recorded_frames = self.record_audio(self.chunk_size) # set param here chunk size
+        return_list = False
+        [recorded_frames, len_of_frames] = self.record_audio(self.chunk_size) # set param here chunk size
         self.wav_list.append(recorded_frames)
-        self.record_count += 1 
-        self.sequence_size = 3 # self.secs/self.chunk_size
+        self.record_count = len(self.wav_list)
+        self.sequence_size = 11 # self.secs/self.chunk_size
         if (self.record_count % self.sequence_size == 0):
-            return_list =  self.wav_list
+            return_list =  True
+            print("length")
+            print(len(self.wav_list))
             print("Collected %d seconds of audio" %(self.secs))
             # Remove fist two objects in array to remove 0.2 seconds of audio data from array
-            self.wav_list.pop(0)
-            self.wav_list.pop(0)
+    
             # send the frames at the beginning of the list (save as wav for now)
-            return return_list
+        return [return_list, len_of_frames]
             
 
 
@@ -243,11 +248,14 @@ class ROSInterface:
 
         frames = []
         arr_length = int(RESPEAKER_RATE / CHUNK * seconds) # get length of array to produce n seconds of audio
+        print("arr length")
+        print(arr_length)
         for i in range(0, arr_length):
             data = stream.read(CHUNK)
             a = np.frombuffer(data,dtype=np.int16)[0::6] # extracts fused channel 0
             self.audio_data_pub.publish(a)
-            frames.append(a.tobytes())
+            # print(a)
+            frames.append(a)
         
         # Check to prevent empty frames list
         if (len(frames) == 0):
@@ -259,7 +267,7 @@ class ROSInterface:
         stream.close()
         p.terminate()
 
-        return frames
+        return [frames, arr_length]
 
 
     def process_audio_loop(self):
@@ -268,14 +276,26 @@ class ROSInterface:
             if dev:
                 while not rospy.is_shutdown():
                     if self.respeaker.is_voice() == 1:
-                        # wav_data = list of lists of bytes
-                        wav_data = self.get_audio()
-                        # if wav_data is a list
-                        if(isinstance(wav_data, list)):
+
+                        [iter, data_len] = self.get_audio()
+
+                        # if the wav list has values
+                        if(iter):
                             # flatten list
-                            flat_list = [item for sublist in wav_data for item in sublist]
+                            flat_list = [item for sublist in self.wav_list for item in sublist]
+                            flatter_list = [item for sublist in flat_list for item in sublist]
+
+                            flatter_list_np = np.asarray(flatter_list)
+
+                            # Test that the data is good
+                            # scipy.io.wavfile.write(self.dir, 16000, flatter_list_np)
+
+                            # remove first 0.2 seconds from wav data
+                            self.wav_list.pop(0)
+                            self.wav_list.pop(0)
+
                             # Call of Utku's function
-                            self.cough_prob = dsp.classify_cough(flat_list,RESPEAKER_RATE,self.loaded_model,self.loaded_scaler)
+                            self.cough_prob = dsp.classify_cough(flatter_list_np,RESPEAKER_RATE,self.loaded_model,self.loaded_scaler)
                             print("%d %% probability of cough" %(self.cough_prob))
                             # do something with probability (publish?)
         except usb.core.USBError: