diff --git a/ivas_processing_scripts/audiotools/__init__.py b/ivas_processing_scripts/audiotools/__init__.py
index f392771dc5b43206c0aac7d4fb08e0d90ce42e9c..7d835d1663a6ed3754f1743ad499f9d9ec5d52be 100755
--- a/ivas_processing_scripts/audiotools/__init__.py
+++ b/ivas_processing_scripts/audiotools/__init__.py
@@ -34,7 +34,10 @@ import argparse
 from itertools import repeat
 from pathlib import Path
 
-from ivas_processing_scripts.audiotools.constants import AUDIO_FORMATS, BINAURAL_LFE_GAIN
+from ivas_processing_scripts.audiotools.constants import (
+    AUDIO_FORMATS,
+    BINAURAL_LFE_GAIN,
+)
 from ivas_processing_scripts.audiotools.convert import convert_file
 from ivas_processing_scripts.utils import apply_func_parallel
 
diff --git a/other/lp16k.py b/other/lp16k.py
index 975356aa07538a9ba5aacba837ecc55f3d101cb7..9f078881bbbecaef5a2623de4329457625bdb176 100755
--- a/other/lp16k.py
+++ b/other/lp16k.py
@@ -7,6 +7,7 @@ from time import sleep
 
 sys.path.append(str(Path(__file__).parent.parent))
 from ivas_processing_scripts.audiotools.audio import fromfile
+from ivas_processing_scripts.audiotools.audioarray import trim
 from ivas_processing_scripts.audiotools.audiofile import write
 from ivas_processing_scripts.audiotools.wrappers.filter import resample_itu
 from ivas_processing_scripts.utils import progressbar_update, spinner
@@ -19,21 +20,24 @@ def lp16k(in_file, out_file):
             f"Unsupported sampling rate {x.fs//1000} kHz for input file {in_file} - only 48 kHz is supported!"
         )
 
+    # save length of audio since resampling can alter the length by one sample due to rounding
+    len_signal = len(x.audio)
+
     # resample ITU only returns the audio array
     # but uses the sampling rate stored in the audio object
     # so we need to correctly set it after each call
-    x.audio = resample_itu(x, 96000)
-    x.fs = 96000
-
     x.audio = resample_itu(x, 32000)
     x.fs = 32000
 
-    x.audio = resample_itu(x, 96000)
-    x.fs = 96000
-
     x.audio = resample_itu(x, 48000)
     x.fs = 48000
 
+    if len_signal != len(x.audio):
+        # padding should usually not happen since the resampling only rounds up
+        x.audio = trim(
+            x.audio, x.fs, (0, len(x.audio) - len_signal), samples=True, pad_noise=True
+        )
+
     write(out_file, x.audio, x.fs)